am e6105c26: (-s ours) Reconcile with jb-mr2-zeroday-release - do not merge
* commit 'e6105c26a5b506f96e9fe1d3fec64a8b2ed1d6dd':
diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index 1b136de..fd78572 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -255,6 +255,14 @@
mCamera->setPreviewCallbackFlag(flag);
}
+status_t Camera::setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer)
+{
+ sp <ICamera> c = mCamera;
+ if (c == 0) return NO_INIT;
+ return c->setPreviewCallbackTarget(callbackProducer);
+}
+
// callback from camera service
void Camera::notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2)
{
diff --git a/camera/ICamera.cpp b/camera/ICamera.cpp
index 8900867..12356f0 100644
--- a/camera/ICamera.cpp
+++ b/camera/ICamera.cpp
@@ -31,6 +31,7 @@
DISCONNECT = IBinder::FIRST_CALL_TRANSACTION,
SET_PREVIEW_TEXTURE,
SET_PREVIEW_CALLBACK_FLAG,
+ SET_PREVIEW_CALLBACK_TARGET,
START_PREVIEW,
STOP_PREVIEW,
AUTO_FOCUS,
@@ -65,6 +66,7 @@
Parcel data, reply;
data.writeInterfaceToken(ICamera::getInterfaceDescriptor());
remote()->transact(DISCONNECT, data, &reply);
+ reply.readExceptionCode();
}
// pass the buffered IGraphicBufferProducer to the camera service
@@ -90,6 +92,18 @@
remote()->transact(SET_PREVIEW_CALLBACK_FLAG, data, &reply);
}
+ status_t setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer)
+ {
+ ALOGV("setPreviewCallbackTarget");
+ Parcel data, reply;
+ data.writeInterfaceToken(ICamera::getInterfaceDescriptor());
+ sp<IBinder> b(callbackProducer->asBinder());
+ data.writeStrongBinder(b);
+ remote()->transact(SET_PREVIEW_CALLBACK_TARGET, data, &reply);
+ return reply.readInt32();
+ }
+
// start preview mode, must call setPreviewDisplay first
status_t startPreview()
{
@@ -268,6 +282,7 @@
ALOGV("DISCONNECT");
CHECK_INTERFACE(ICamera, data, reply);
disconnect();
+ reply->writeNoException();
return NO_ERROR;
} break;
case SET_PREVIEW_TEXTURE: {
@@ -285,6 +300,14 @@
setPreviewCallbackFlag(callback_flag);
return NO_ERROR;
} break;
+ case SET_PREVIEW_CALLBACK_TARGET: {
+ ALOGV("SET_PREVIEW_CALLBACK_TARGET");
+ CHECK_INTERFACE(ICamera, data, reply);
+ sp<IGraphicBufferProducer> cp =
+ interface_cast<IGraphicBufferProducer>(data.readStrongBinder());
+ reply->writeInt32(setPreviewCallbackTarget(cp));
+ return NO_ERROR;
+ }
case START_PREVIEW: {
ALOGV("START_PREVIEW");
CHECK_INTERFACE(ICamera, data, reply);
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index 134f7f0..819e410 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -15,6 +15,9 @@
** limitations under the License.
*/
+#define LOG_TAG "BpCameraService"
+#include <utils/Log.h>
+
#include <stdint.h>
#include <sys/types.h>
@@ -31,6 +34,53 @@
namespace android {
+namespace {
+
+enum {
+ EX_SECURITY = -1,
+ EX_BAD_PARCELABLE = -2,
+ EX_ILLEGAL_ARGUMENT = -3,
+ EX_NULL_POINTER = -4,
+ EX_ILLEGAL_STATE = -5,
+ EX_HAS_REPLY_HEADER = -128, // special; see below
+};
+
+static bool readExceptionCode(Parcel& reply) {
+ int32_t exceptionCode = reply.readExceptionCode();
+
+ if (exceptionCode != 0) {
+ const char* errorMsg;
+ switch(exceptionCode) {
+ case EX_SECURITY:
+ errorMsg = "Security";
+ break;
+ case EX_BAD_PARCELABLE:
+ errorMsg = "BadParcelable";
+ break;
+ case EX_NULL_POINTER:
+ errorMsg = "NullPointer";
+ break;
+ case EX_ILLEGAL_STATE:
+ errorMsg = "IllegalState";
+ break;
+ // Binder should be handling this code inside Parcel::readException
+ // but lets have a to-string here anyway just in case.
+ case EX_HAS_REPLY_HEADER:
+ errorMsg = "HasReplyHeader";
+ break;
+ default:
+ errorMsg = "Unknown";
+ }
+
+ ALOGE("Binder transmission error %s (%d)", errorMsg, exceptionCode);
+ return true;
+ }
+
+ return false;
+}
+
+};
+
class BpCameraService: public BpInterface<ICameraService>
{
public:
@@ -45,6 +95,8 @@
Parcel data, reply;
data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
remote()->transact(BnCameraService::GET_NUMBER_OF_CAMERAS, data, &reply);
+
+ if (readExceptionCode(reply)) return 0;
return reply.readInt32();
}
@@ -55,9 +107,14 @@
data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
data.writeInt32(cameraId);
remote()->transact(BnCameraService::GET_CAMERA_INFO, data, &reply);
- cameraInfo->facing = reply.readInt32();
- cameraInfo->orientation = reply.readInt32();
- return reply.readInt32();
+
+ if (readExceptionCode(reply)) return -EPROTO;
+ status_t result = reply.readInt32();
+ if (reply.readInt32() != 0) {
+ cameraInfo->facing = reply.readInt32();
+ cameraInfo->orientation = reply.readInt32();
+ }
+ return result;
}
// connect to camera service
@@ -71,6 +128,8 @@
data.writeString16(clientPackageName);
data.writeInt32(clientUid);
remote()->transact(BnCameraService::CONNECT, data, &reply);
+
+ if (readExceptionCode(reply)) return NULL;
return interface_cast<ICamera>(reply.readStrongBinder());
}
@@ -85,6 +144,8 @@
data.writeString16(clientPackageName);
data.writeInt32(clientUid);
remote()->transact(BnCameraService::CONNECT_PRO, data, &reply);
+
+ if (readExceptionCode(reply)) return NULL;
return interface_cast<IProCameraUser>(reply.readStrongBinder());
}
@@ -94,6 +155,8 @@
data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
data.writeStrongBinder(listener->asBinder());
remote()->transact(BnCameraService::ADD_LISTENER, data, &reply);
+
+ if (readExceptionCode(reply)) return -EPROTO;
return reply.readInt32();
}
@@ -103,6 +166,8 @@
data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
data.writeStrongBinder(listener->asBinder());
remote()->transact(BnCameraService::REMOVE_LISTENER, data, &reply);
+
+ if (readExceptionCode(reply)) return -EPROTO;
return reply.readInt32();
}
};
@@ -117,17 +182,22 @@
switch(code) {
case GET_NUMBER_OF_CAMERAS: {
CHECK_INTERFACE(ICameraService, data, reply);
+ reply->writeNoException();
reply->writeInt32(getNumberOfCameras());
return NO_ERROR;
} break;
case GET_CAMERA_INFO: {
CHECK_INTERFACE(ICameraService, data, reply);
- CameraInfo cameraInfo;
+ CameraInfo cameraInfo = CameraInfo();
memset(&cameraInfo, 0, sizeof(cameraInfo));
status_t result = getCameraInfo(data.readInt32(), &cameraInfo);
+ reply->writeNoException();
+ reply->writeInt32(result);
+
+ // Fake a parcelable object here
+ reply->writeInt32(1); // means the parcelable is included
reply->writeInt32(cameraInfo.facing);
reply->writeInt32(cameraInfo.orientation);
- reply->writeInt32(result);
return NO_ERROR;
} break;
case CONNECT: {
@@ -139,17 +209,20 @@
int32_t clientUid = data.readInt32();
sp<ICamera> camera = connect(cameraClient, cameraId,
clientName, clientUid);
+ reply->writeNoException();
reply->writeStrongBinder(camera->asBinder());
return NO_ERROR;
} break;
case CONNECT_PRO: {
CHECK_INTERFACE(ICameraService, data, reply);
- sp<IProCameraCallbacks> cameraClient = interface_cast<IProCameraCallbacks>(data.readStrongBinder());
+ sp<IProCameraCallbacks> cameraClient =
+ interface_cast<IProCameraCallbacks>(data.readStrongBinder());
int32_t cameraId = data.readInt32();
const String16 clientName = data.readString16();
int32_t clientUid = data.readInt32();
sp<IProCameraUser> camera = connect(cameraClient, cameraId,
clientName, clientUid);
+ reply->writeNoException();
reply->writeStrongBinder(camera->asBinder());
return NO_ERROR;
} break;
@@ -157,6 +230,7 @@
CHECK_INTERFACE(ICameraService, data, reply);
sp<ICameraServiceListener> listener =
interface_cast<ICameraServiceListener>(data.readStrongBinder());
+ reply->writeNoException();
reply->writeInt32(addListener(listener));
return NO_ERROR;
} break;
@@ -164,6 +238,7 @@
CHECK_INTERFACE(ICameraService, data, reply);
sp<ICameraServiceListener> listener =
interface_cast<ICameraServiceListener>(data.readStrongBinder());
+ reply->writeNoException();
reply->writeInt32(removeListener(listener));
return NO_ERROR;
} break;
diff --git a/camera/ICameraServiceListener.cpp b/camera/ICameraServiceListener.cpp
index 640ee35..b2f1729 100644
--- a/camera/ICameraServiceListener.cpp
+++ b/camera/ICameraServiceListener.cpp
@@ -54,6 +54,8 @@
data,
&reply,
IBinder::FLAG_ONEWAY);
+
+ reply.readExceptionCode();
}
};
@@ -73,6 +75,7 @@
int32_t cameraId = data.readInt32();
onStatusChanged(status, cameraId);
+ reply->writeNoException();
return NO_ERROR;
} break;
diff --git a/camera/IProCameraUser.cpp b/camera/IProCameraUser.cpp
index 4c4dec3..015cb5c 100644
--- a/camera/IProCameraUser.cpp
+++ b/camera/IProCameraUser.cpp
@@ -162,6 +162,7 @@
Parcel data, reply;
data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
remote()->transact(DISCONNECT, data, &reply);
+ reply.readExceptionCode();
}
virtual status_t connect(const sp<IProCameraCallbacks>& cameraClient)
@@ -307,6 +308,7 @@
ALOGV("DISCONNECT");
CHECK_INTERFACE(IProCameraUser, data, reply);
disconnect();
+ reply->writeNoException();
return NO_ERROR;
} break;
case CONNECT: {
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 3844487..1060131 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -19,8 +19,6 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
-
LOCAL_MODULE:= stagefright
include $(BUILD_EXECUTABLE)
@@ -42,7 +40,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= record
@@ -65,7 +63,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= recordvideo
@@ -89,7 +87,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= audioloop
@@ -112,7 +110,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= stream
@@ -135,7 +133,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= sf2
@@ -159,7 +157,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= codec
@@ -182,7 +180,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= muxer
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 115b07c..924cf6d 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -30,8 +30,6 @@
#include <binder/ProcessState.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include "include/LiveSession.h"
#include "include/NuCachedSource2.h"
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/DataSource.h>
@@ -678,7 +676,6 @@
gDisplayHistogram = false;
sp<ALooper> looper;
- sp<LiveSession> liveSession;
int res;
while ((res = getopt(argc, argv, "han:lm:b:ptsrow:kxSTd:D:")) >= 0) {
@@ -961,9 +958,7 @@
sp<DataSource> dataSource = DataSource::CreateFromURI(filename);
- if (strncasecmp(filename, "sine:", 5)
- && strncasecmp(filename, "httplive://", 11)
- && dataSource == NULL) {
+ if (strncasecmp(filename, "sine:", 5) && dataSource == NULL) {
fprintf(stderr, "Unable to create data source.\n");
return 1;
}
@@ -995,44 +990,21 @@
mediaSources.push(mediaSource);
}
} else {
- sp<MediaExtractor> extractor;
+ sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
- if (!strncasecmp("httplive://", filename, 11)) {
- String8 uri("http://");
- uri.append(filename + 11);
+ if (extractor == NULL) {
+ fprintf(stderr, "could not create extractor.\n");
+ return -1;
+ }
- if (looper == NULL) {
- looper = new ALooper;
- looper->start();
- }
- liveSession = new LiveSession(NULL /* notify */);
- looper->registerHandler(liveSession);
+ sp<MetaData> meta = extractor->getMetaData();
- liveSession->connect(uri.string());
- dataSource = liveSession->getDataSource();
+ if (meta != NULL) {
+ const char *mime;
+ CHECK(meta->findCString(kKeyMIMEType, &mime));
- extractor =
- MediaExtractor::Create(
- dataSource, MEDIA_MIMETYPE_CONTAINER_MPEG2TS);
-
- syncInfoPresent = false;
- } else {
- extractor = MediaExtractor::Create(dataSource);
-
- if (extractor == NULL) {
- fprintf(stderr, "could not create extractor.\n");
- return -1;
- }
-
- sp<MetaData> meta = extractor->getMetaData();
-
- if (meta != NULL) {
- const char *mime;
- CHECK(meta->findCString(kKeyMIMEType, &mime));
-
- if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2TS)) {
- syncInfoPresent = false;
- }
+ if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2TS)) {
+ syncInfoPresent = false;
}
}
diff --git a/include/camera/Camera.h b/include/camera/Camera.h
index 37626a4..c34b3ea 100644
--- a/include/camera/Camera.h
+++ b/include/camera/Camera.h
@@ -121,7 +121,15 @@
void setListener(const sp<CameraListener>& listener);
void setRecordingProxyListener(const sp<ICameraRecordingProxyListener>& listener);
+
+ // Configure preview callbacks to app. Only one of the older
+ // callbacks or the callback surface can be active at the same time;
+ // enabling one will disable the other if active. Flags can be
+ // disabled by calling it with CAMERA_FRAME_CALLBACK_FLAG_NOOP, and
+ // Target by calling it with a NULL interface.
void setPreviewCallbackFlags(int preview_callback_flag);
+ status_t setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer);
sp<ICameraRecordingProxy> getRecordingProxy();
diff --git a/include/camera/ICamera.h b/include/camera/ICamera.h
index 2236c1f..f3a186e 100644
--- a/include/camera/ICamera.h
+++ b/include/camera/ICamera.h
@@ -32,6 +32,9 @@
class ICamera: public IInterface
{
+ /**
+ * Keep up-to-date with ICamera.aidl in frameworks/base
+ */
public:
DECLARE_META_INTERFACE(Camera);
@@ -51,8 +54,15 @@
const sp<IGraphicBufferProducer>& bufferProducer) = 0;
// set the preview callback flag to affect how the received frames from
- // preview are handled.
+ // preview are handled. Enabling preview callback flags disables any active
+ // preview callback surface set by setPreviewCallbackTarget().
virtual void setPreviewCallbackFlag(int flag) = 0;
+ // set a buffer interface to use for client-received preview frames instead
+ // of preview callback buffers. Passing a valid interface here disables any
+ // active preview callbacks set by setPreviewCallbackFlag(). Passing NULL
+ // disables the use of the callback target.
+ virtual status_t setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer) = 0;
// start preview mode, must call setPreviewDisplay first
virtual status_t startPreview() = 0;
diff --git a/include/camera/ICameraClient.h b/include/camera/ICameraClient.h
index b30aa7a..1584dba 100644
--- a/include/camera/ICameraClient.h
+++ b/include/camera/ICameraClient.h
@@ -28,6 +28,9 @@
class ICameraClient: public IInterface
{
+ /**
+ * Keep up-to-date with ICameraClient.aidl in frameworks/base
+ */
public:
DECLARE_META_INTERFACE(CameraClient);
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index aaf6eb3..3c2e60a 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -32,6 +32,9 @@
class ICameraService : public IInterface
{
public:
+ /**
+ * Keep up-to-date with ICameraService.aidl in frameworks/base
+ */
enum {
GET_NUMBER_OF_CAMERAS = IBinder::FIRST_CALL_TRANSACTION,
GET_CAMERA_INFO,
diff --git a/include/camera/ICameraServiceListener.h b/include/camera/ICameraServiceListener.h
index f2a11c2..0a0e43a 100644
--- a/include/camera/ICameraServiceListener.h
+++ b/include/camera/ICameraServiceListener.h
@@ -26,6 +26,9 @@
class ICameraServiceListener : public IInterface
{
+ /**
+ * Keep up-to-date with ICameraServiceListener.aidl in frameworks/base
+ */
public:
/**
diff --git a/include/camera/IProCameraCallbacks.h b/include/camera/IProCameraCallbacks.h
index 563ec17..c774698 100644
--- a/include/camera/IProCameraCallbacks.h
+++ b/include/camera/IProCameraCallbacks.h
@@ -30,6 +30,9 @@
class IProCameraCallbacks : public IInterface
{
+ /**
+ * Keep up-to-date with IProCameraCallbacks.aidl in frameworks/base
+ */
public:
DECLARE_META_INTERFACE(ProCameraCallbacks);
diff --git a/include/camera/IProCameraUser.h b/include/camera/IProCameraUser.h
index 45b818c..2ccc4d2 100644
--- a/include/camera/IProCameraUser.h
+++ b/include/camera/IProCameraUser.h
@@ -34,6 +34,9 @@
class IProCameraUser: public IInterface
{
+ /**
+ * Keep up-to-date with IProCameraUser.aidl in frameworks/base
+ */
public:
DECLARE_META_INTERFACE(ProCameraUser);
diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
index 43e4de7..ef392f0 100644
--- a/include/media/AudioBufferProvider.h
+++ b/include/media/AudioBufferProvider.h
@@ -26,6 +26,8 @@
{
public:
+ // FIXME merge with AudioTrackShared::Buffer, AudioTrack::Buffer, and AudioRecord::Buffer
+ // and rename getNextBuffer() to obtainBuffer()
struct Buffer {
Buffer() : raw(NULL), frameCount(0) { }
union {
@@ -44,6 +46,19 @@
// pts is the local time when the next sample yielded by getNextBuffer
// will be rendered.
// Pass kInvalidPTS if the PTS is unknown or not applicable.
+ // On entry:
+ // buffer != NULL
+ // buffer->raw unused
+ // buffer->frameCount maximum number of desired frames
+ // On successful return:
+ // status NO_ERROR
+ // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames
+ // buffer->frameCount number of contiguous available frames at buffer->raw,
+ // 0 < buffer->frameCount <= entry value
+ // On error return:
+ // status != NO_ERROR
+ // buffer->raw NULL
+ // buffer->frameCount 0
virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0;
virtual void releaseBuffer(Buffer* buffer) = 0;
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 38c6548..7aa3c24 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -14,26 +14,24 @@
* limitations under the License.
*/
-#ifndef AUDIORECORD_H_
-#define AUDIORECORD_H_
+#ifndef ANDROID_AUDIORECORD_H
+#define ANDROID_AUDIORECORD_H
-#include <binder/IMemory.h>
#include <cutils/sched_policy.h>
#include <media/AudioSystem.h>
#include <media/IAudioRecord.h>
-#include <system/audio.h>
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
#include <utils/threads.h>
namespace android {
+// ----------------------------------------------------------------------------
+
class audio_track_cblk_t;
class AudioRecordClientProxy;
// ----------------------------------------------------------------------------
-class AudioRecord : virtual public RefBase
+class AudioRecord : public RefBase
{
public:
@@ -49,6 +47,8 @@
// (See setMarkerPosition()).
EVENT_NEW_POS = 3, // Record head is at a new position
// (See setPositionUpdatePeriod()).
+ EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and
+ // voluntary invalidation by mediaserver, or mediaserver crash.
};
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -58,11 +58,16 @@
class Buffer
{
public:
+ // FIXME use m prefix
size_t frameCount; // number of sample frames corresponding to size;
// on input it is the number of frames available,
// on output is the number of frames actually drained
- size_t size; // total size in bytes == frameCount * frameSize
+ size_t size; // input/output in bytes == frameCount * frameSize
+ // FIXME this is redundant with respect to frameCount,
+ // and TRANSFER_OBTAIN mode is broken for 8-bit data
+ // since we don't define the frame format
+
union {
void* raw;
short* i16; // signed 16-bit
@@ -84,6 +89,7 @@
* - EVENT_OVERRUN: unused.
* - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
* - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
+ * - EVENT_NEW_IAUDIORECORD: unused.
*/
typedef void (*callback_t)(int event, void* user, void *info);
@@ -101,20 +107,28 @@
audio_format_t format,
audio_channel_mask_t channelMask);
+ /* How data is transferred from AudioRecord
+ */
+ enum transfer_type {
+ TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters
+ TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
+ TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
+ TRANSFER_SYNC, // synchronous read()
+ };
+
/* Constructs an uninitialized AudioRecord. No connection with
- * AudioFlinger takes place.
+ * AudioFlinger takes place. Use set() after this.
*/
AudioRecord();
/* Creates an AudioRecord object and registers it with AudioFlinger.
* Once created, the track needs to be started before it can be used.
- * Unspecified values are set to the audio hardware's current
- * values.
+ * Unspecified values are set to appropriate default values.
*
* Parameters:
*
- * inputSource: Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT).
- * sampleRate: Track sampling rate in Hz.
+ * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT).
+ * sampleRate: Data sink sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
* channelMask: Channel mask.
@@ -124,11 +138,13 @@
* be larger if the requested size is not compatible with current audio HAL
* latency. Zero means to use a default value.
* cbf: Callback function. If not null, this function is called periodically
- * to consume new PCM data.
+ * to consume new PCM data and inform of marker, position updates, etc.
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames are ready in record track output buffer.
* sessionId: Not yet supported.
+ * transferType: How data is transferred from AudioRecord.
+ * threadCanCallJava: Not present in parameter list, and so is fixed at false.
*/
AudioRecord(audio_source_t inputSource,
@@ -139,22 +155,28 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0);
-
+ int sessionId = 0,
+ transfer_type transferType = TRANSFER_DEFAULT);
/* Terminates the AudioRecord and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioRecord.
*/
- ~AudioRecord();
+protected:
+ virtual ~AudioRecord();
+public:
-
- /* Initialize an uninitialized AudioRecord.
+ /* Initialize an AudioRecord that was created using the AudioRecord() constructor.
+ * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful intialization
- * - INVALID_OPERATION: AudioRecord is already intitialized or record device is already in use
+ * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
* - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
+ *
+ * Parameters not listed in the AudioRecord constructors above:
+ *
+ * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI.
*/
status_t set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT,
uint32_t sampleRate = 0,
@@ -165,30 +187,29 @@
void* user = NULL,
int notificationFrames = 0,
bool threadCanCallJava = false,
- int sessionId = 0);
-
+ int sessionId = 0,
+ transfer_type transferType = TRANSFER_DEFAULT);
/* Result of constructing the AudioRecord. This must be checked
* before using any AudioRecord API (except for set()), because using
* an uninitialized AudioRecord produces undefined results.
* See set() method above for possible return codes.
*/
- status_t initCheck() const;
+ status_t initCheck() const { return mStatus; }
/* Returns this track's estimated latency in milliseconds.
* This includes the latency due to AudioRecord buffer size,
* and audio hardware driver.
*/
- uint32_t latency() const;
+ uint32_t latency() const { return mLatency; }
/* getters, see constructor and set() */
- audio_format_t format() const;
- uint32_t channelCount() const;
- size_t frameCount() const;
- size_t frameSize() const { return mFrameSize; }
- audio_source_t inputSource() const;
-
+ audio_format_t format() const { return mFormat; }
+ uint32_t channelCount() const { return mChannelCount; }
+ size_t frameCount() const { return mFrameCount; }
+ size_t frameSize() const { return mFrameSize; }
+ audio_source_t inputSource() const { return mInputSource; }
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
@@ -198,26 +219,29 @@
status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
- /* Stop a track. If set, the callback will cease being called and
- * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
- * and will drain buffers until the pool is exhausted.
+ /* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still
+ * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
*/
void stop();
bool stopped() const;
- /* Get sample rate for this record track in Hz.
+ /* Return the sink sample rate for this record track in Hz.
+ * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock.
*/
- uint32_t getSampleRate() const;
+ uint32_t getSampleRate() const { return mSampleRate; }
/* Sets marker position. When record reaches the number of frames specified,
* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
* with marker == 0 cancels marker notification callback.
+ * To set a marker at a position which would compute as 0,
+ * a workaround is to the set the marker at a nearby position such as ~0 or 1.
* If the AudioRecord has been opened with no callback function associated,
* the operation will fail.
*
* Parameters:
*
- * marker: marker position expressed in frames.
+ * marker: marker position expressed in wrapping (overflow) frame units,
+ * like the return value of getPosition().
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
@@ -226,13 +250,13 @@
status_t setMarkerPosition(uint32_t marker);
status_t getMarkerPosition(uint32_t *marker) const;
-
/* Sets position update period. Every time the number of frames specified has been recorded,
* a callback with event type EVENT_NEW_POS is called.
* Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
* callback.
* If the AudioRecord has been opened with no callback function associated,
* the operation will fail.
+ * Extremely small values may be rounded up to a value the implementation can support.
*
* Parameters:
*
@@ -245,13 +269,13 @@
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
-
- /* Gets record head position. The position is the total number of frames
- * recorded since record start.
+ /* Return the total number of frames recorded since recording started.
+ * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz.
+ * It is reset to zero by stop().
*
* Parameters:
*
- * position: Address where to return record head position within AudioRecord buffer.
+ * position: Address where to return record head position.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
@@ -276,38 +300,70 @@
*
* Returned value:
* AudioRecord session ID.
+ *
+ * No lock needed because session ID doesn't change after first set().
*/
- int getSessionId() const;
+ int getSessionId() const { return mSessionId; }
- /* Obtains a buffer of "frameCount" frames. The buffer must be
- * drained entirely, and then released with releaseBuffer().
- * If the track is stopped, obtainBuffer() returns
- * STOPPED instead of NO_ERROR as long as there are buffers available,
- * at which point NO_MORE_BUFFERS is returned.
+ /* Obtains a buffer of up to "audioBuffer->frameCount" full frames.
+ * After draining these frames of data, the caller should release them with releaseBuffer().
+ * If the track buffer is not empty, obtainBuffer() returns as many contiguous
+ * full frames as are available immediately.
+ * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK
+ * regardless of the value of waitCount.
+ * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a
+ * maximum timeout based on waitCount; see chart below.
* Buffers will be returned until the pool
* is exhausted, at which point obtainBuffer() will either block
- * or return WOULD_BLOCK depending on the value of the "blocking"
+ * or return WOULD_BLOCK depending on the value of the "waitCount"
* parameter.
*
+ * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
+ * which should use read() or callback EVENT_MORE_DATA instead.
+ *
* Interpretation of waitCount:
* +n limits wait time to n * WAIT_PERIOD_MS,
* -1 causes an (almost) infinite wait time,
* 0 non-blocking.
+ *
+ * Buffer fields
+ * On entry:
+ * frameCount number of frames requested
+ * After error return:
+ * frameCount 0
+ * size 0
+ * raw undefined
+ * After successful return:
+ * frameCount actual number of frames available, <= number requested
+ * size actual number of bytes available
+ * raw pointer to the buffer
*/
- enum {
- NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
- STOPPED = 1
- };
+ /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
+ status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+ __attribute__((__deprecated__));
- status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+private:
+ /* New internal API.
+ * If nonContig is non-NULL, it is an output parameter that will be set to the number of
+ * additional non-contiguous frames that are available immediately.
+ * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(),
+ * in case the requested amount of frames is in two or more non-contiguous regions.
+ * FIXME requested and elapsed are both relative times. Consider changing to absolute time.
+ */
+ status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
+ struct timespec *elapsed = NULL, size_t *nonContig = NULL);
+public:
- /* Release an emptied buffer of "frameCount" frames for AudioFlinger to re-fill. */
+ /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */
+ // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
void releaseBuffer(Buffer* audioBuffer);
-
/* As a convenience we provide a read() interface to the audio buffer.
- * This is implemented on top of obtainBuffer/releaseBuffer.
+ * Input parameter 'size' is in byte units.
+ * This is implemented on top of obtainBuffer/releaseBuffer. For best
+ * performance use callbacks. Returns actual number of bytes read >= 0,
+ * or a negative status code.
*/
ssize_t read(void* buffer, size_t size);
@@ -336,68 +392,113 @@
void pause(); // suspend thread from execution at next loop boundary
void resume(); // allow thread to execute, if not requested to exit
+ void pauseConditional();
+ // like pause(), but only if prior resume() wasn't latched
private:
friend class AudioRecord;
virtual bool threadLoop();
- AudioRecord& mReceiver;
+ AudioRecord& mReceiver;
virtual ~AudioRecordThread();
Mutex mMyLock; // Thread::mLock is private
Condition mMyCond; // Thread::mThreadExitedCondition is private
bool mPaused; // whether thread is currently paused
+ bool mResumeLatch; // whether next pauseConditional() will be a nop
};
// body of AudioRecordThread::threadLoop()
- bool processAudioBuffer(const sp<AudioRecordThread>& thread);
+ // returns the maximum amount of time before we would like to run again, where:
+ // 0 immediately
+ // > 0 no later than this many nanoseconds from now
+ // NS_WHENEVER still active but no particular deadline
+ // NS_INACTIVE inactive so don't run again until re-started
+ // NS_NEVER never again
+ static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
+ nsecs_t processAudioBuffer(const sp<AudioRecordThread>& thread);
+ // caller must hold lock on mLock for all _l methods
status_t openRecord_l(uint32_t sampleRate,
audio_format_t format,
size_t frameCount,
- audio_io_handle_t input);
+ audio_io_handle_t input,
+ size_t epoch);
+
audio_io_handle_t getInput_l();
- status_t restoreRecord_l(audio_track_cblk_t*& cblk);
+
+ // FIXME enum is faster than strcmp() for parameter 'from'
+ status_t restoreRecord_l(const char *from);
sp<AudioRecordThread> mAudioRecordThread;
mutable Mutex mLock;
- bool mActive; // protected by mLock
+ // Current client state: false = stopped, true = active. Protected by mLock. If more states
+ // are added, consider changing this to enum State { ... } mState as in AudioTrack.
+ bool mActive;
// for client callback handler
callback_t mCbf; // callback handler for events, or NULL
- void* mUserData;
+ void* mUserData; // for client callback handler
// for notification APIs
- uint32_t mNotificationFrames;
- uint32_t mRemainingFrames;
- uint32_t mMarkerPosition; // in frames
+ uint32_t mNotificationFrames; // frames between each notification callback
+ bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+
+ // These are private to processAudioBuffer(), and are not protected by a lock
+ uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
+ bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
+ int mObservedSequence; // last observed value of mSequence
+
+ uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
uint32_t mNewPosition; // in frames
- uint32_t mUpdatePeriod; // in ms
+ uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
+
+ status_t mStatus;
// constant after constructor or set()
uint32_t mSampleRate;
size_t mFrameCount;
audio_format_t mFormat;
- uint8_t mChannelCount;
+ uint32_t mChannelCount;
size_t mFrameSize; // app-level frame size == AudioFlinger frame size
audio_source_t mInputSource;
- status_t mStatus;
- uint32_t mLatency;
+ uint32_t mLatency; // in ms
audio_channel_mask_t mChannelMask;
- audio_io_handle_t mInput; // returned by AudioSystem::getInput()
int mSessionId;
+ transfer_type mTransfer;
+
+ audio_io_handle_t mInput; // returned by AudioSystem::getInput()
// may be changed if IAudioRecord object is re-created
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
- audio_track_cblk_t* mCblk;
- void* mBuffers; // starting address of buffers in shared memory
+ audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
- int mPreviousPriority; // before start()
+ int mPreviousPriority; // before start()
SchedPolicy mPreviousSchedulingGroup;
- AudioRecordClientProxy* mProxy;
+
+ // The proxy should only be referenced while a lock is held because the proxy isn't
+ // multi-thread safe.
+ // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock,
+ // provided that the caller also holds an extra reference to the proxy and shared memory to keep
+ sp<AudioRecordClientProxy> mProxy;
+
+ bool mInOverrun; // whether recorder is currently in overrun state
+
+private:
+ class DeathNotifier : public IBinder::DeathRecipient {
+ public:
+ DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { }
+ protected:
+ virtual void binderDied(const wp<IBinder>& who);
+ private:
+ const wp<AudioRecord> mAudioRecord;
+ };
+
+ sp<DeathNotifier> mDeathNotifier;
+ uint32_t mSequence; // incremented for each new IAudioRecord attempt
};
}; // namespace android
-#endif /*AUDIORECORD_H_*/
+#endif // ANDROID_AUDIORECORD_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 64f82bb..e9bb76a 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -17,18 +17,9 @@
#ifndef ANDROID_AUDIOTRACK_H
#define ANDROID_AUDIOTRACK_H
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <media/IAudioFlinger.h>
-#include <media/IAudioTrack.h>
-#include <media/AudioSystem.h>
-
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
-#include <binder/IInterface.h>
-#include <binder/IMemory.h>
#include <cutils/sched_policy.h>
+#include <media/AudioSystem.h>
+#include <media/IAudioTrack.h>
#include <utils/threads.h>
namespace android {
@@ -37,10 +28,11 @@
class audio_track_cblk_t;
class AudioTrackClientProxy;
+class StaticAudioTrackClientProxy;
// ----------------------------------------------------------------------------
-class AudioTrack : virtual public RefBase
+class AudioTrack : public RefBase
{
public:
enum channel_index {
@@ -49,7 +41,7 @@
RIGHT = 1
};
- /* Events used by AudioTrack callback function (audio_track_cblk_t).
+ /* Events used by AudioTrack callback function (callback_t).
* Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*.
*/
enum event_type {
@@ -64,7 +56,10 @@
// (See setMarkerPosition()).
EVENT_NEW_POS = 4, // Playback head is at a new position
// (See setPositionUpdatePeriod()).
- EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer.
+ EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer.
+ // Not currently used by android.media.AudioTrack.
+ EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and
+ // voluntary invalidation by mediaserver, or mediaserver crash.
};
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -74,19 +69,23 @@
class Buffer
{
public:
+ // FIXME use m prefix
size_t frameCount; // number of sample frames corresponding to size;
// on input it is the number of frames desired,
// on output is the number of frames actually filled
- size_t size; // input/output in byte units
+ size_t size; // input/output in bytes == frameCount * frameSize
+ // FIXME this is redundant with respect to frameCount,
+ // and TRANSFER_OBTAIN mode is broken for 8-bit data
+ // since we don't define the frame format
+
union {
void* raw;
- short* i16; // signed 16-bit
- int8_t* i8; // unsigned 8-bit, offset by 0x80
+ short* i16; // signed 16-bit
+ int8_t* i8; // unsigned 8-bit, offset by 0x80
};
};
-
/* As a convenience, if a callback is supplied, a handler thread
* is automatically created with the appropriate priority. This thread
* invokes the callback when a new buffer becomes available or various conditions occur.
@@ -100,9 +99,10 @@
* written.
* - EVENT_UNDERRUN: unused.
* - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
- * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
- * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
+ * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
+ * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
* - EVENT_BUFFER_END: unused.
+ * - EVENT_NEW_IAUDIOTRACK: unused.
*/
typedef void (*callback_t)(int event, void* user, void *info);
@@ -114,9 +114,19 @@
* - NO_INIT: audio server or audio hardware not initialized
*/
- static status_t getMinFrameCount(size_t* frameCount,
- audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
- uint32_t sampleRate = 0);
+ static status_t getMinFrameCount(size_t* frameCount,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate);
+
+ /* How data is transferred to AudioTrack
+ */
+ enum transfer_type {
+ TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters
+ TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
+ TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
+ TRANSFER_SYNC, // synchronous write()
+ TRANSFER_SHARED, // shared memory
+ };
/* Constructs an uninitialized AudioTrack. No connection with
* AudioFlinger takes place. Use set() after this.
@@ -128,13 +138,13 @@
* Unspecified values are set to appropriate default values.
* With this constructor, the track is configured for streaming mode.
* Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA.
- * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is deprecated.
+ * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed.
*
* Parameters:
*
* streamType: Select the type of audio stream this track is attached to
* (e.g. AUDIO_STREAM_MUSIC).
- * sampleRate: Track sampling rate in Hz.
+ * sampleRate: Data source sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
* channelMask: Channel mask.
@@ -149,9 +159,10 @@
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames have been consumed from track input buffer.
+ * This is expressed in units of frames at the initial source sample rate.
* sessionId: Specific session ID, or zero to use default.
- * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI.
- * If not present in parameter list, then fixed at false.
+ * transferType: How data is transferred to AudioTrack.
+ * threadCanCallJava: Not present in parameter list, and so is fixed at false.
*/
AudioTrack( audio_stream_type_t streamType,
@@ -163,7 +174,8 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0);
+ int sessionId = 0,
+ transfer_type transferType = TRANSFER_DEFAULT);
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
@@ -174,7 +186,6 @@
* The write() method is not supported in this case.
* It is recommended to pass a callback function to be notified of playback end by an
* EVENT_UNDERRUN event.
- * FIXME EVENT_MORE_DATA still occurs; it must be ignored.
*/
AudioTrack( audio_stream_type_t streamType,
@@ -186,14 +197,18 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0);
+ int sessionId = 0,
+ transfer_type transferType = TRANSFER_DEFAULT);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
*/
- ~AudioTrack();
+protected:
+ virtual ~AudioTrack();
+public:
- /* Initialize an uninitialized AudioTrack.
+ /* Initialize an AudioTrack that was created using the AudioTrack() constructor.
+ * Don't call set() more than once, or after the AudioTrack() constructors that take parameters.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful initialization
* - INVALID_OPERATION: AudioTrack is already initialized
@@ -201,6 +216,10 @@
* - NO_INIT: audio server or audio hardware not initialized
* If sharedBuffer is non-0, the frameCount parameter is ignored and
* replaced by the shared buffer's total allocated size in frame units.
+ *
+ * Parameters not listed in the AudioTrack constructors above:
+ *
+ * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI.
*/
status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
uint32_t sampleRate = 0,
@@ -213,7 +232,8 @@
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
- int sessionId = 0);
+ int sessionId = 0,
+ transfer_type transferType = TRANSFER_DEFAULT);
/* Result of constructing the AudioTrack. This must be checked
* before using any AudioTrack API (except for set()), because using
@@ -233,15 +253,16 @@
audio_stream_type_t streamType() const { return mStreamType; }
audio_format_t format() const { return mFormat; }
- /* Return frame size in bytes, which for linear PCM is channelCount * (bit depth per channel / 8).
+ /* Return frame size in bytes, which for linear PCM is
+ * channelCount * (bit depth per channel / 8).
* channelCount is determined from channelMask, and bit depth comes from format.
* For non-linear formats, the frame size is typically 1 byte.
*/
- uint32_t channelCount() const { return mChannelCount; }
-
- uint32_t frameCount() const { return mFrameCount; }
size_t frameSize() const { return mFrameSize; }
+ uint32_t channelCount() const { return mChannelCount; }
+ uint32_t frameCount() const { return mFrameCount; }
+
/* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
@@ -253,10 +274,9 @@
/* Stop a track.
* In static buffer mode, the track is stopped immediately.
- * In streaming mode, the callback will cease being called and
- * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
- * and will fill up buffers until the pool is exhausted.
- * The stop does not occur immediately: any data remaining in the buffer
+ * In streaming mode, the callback will cease being called. Note that obtainBuffer() still
+ * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK.
+ * In streaming mode the stop does not occur immediately: any data remaining in the buffer
* is first drained, mixed, and output, and only then is the track marked as stopped.
*/
void stop();
@@ -270,7 +290,7 @@
void flush();
/* Pause a track. After pause, the callback will cease being called and
- * obtainBuffer returns STOPPED. Note that obtainBuffer() still works
+ * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works
* and will fill up buffers until the pool is exhausted.
* Volume is ramped down over the next mix buffer following the pause request,
* and then the track is marked as paused. It can be resumed with ramp up by start().
@@ -294,32 +314,41 @@
status_t setAuxEffectSendLevel(float level);
void getAuxEffectSendLevel(float* level) const;
- /* Set sample rate for this track in Hz, mostly used for games' sound effects
+ /* Set source sample rate for this track in Hz, mostly used for games' sound effects
*/
status_t setSampleRate(uint32_t sampleRate);
- /* Return current sample rate in Hz, or 0 if unknown */
+ /* Return current source sample rate in Hz, or 0 if unknown */
uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
* Only supported for static buffer mode.
*
+ * FIXME The comments below are for the new planned interpretation which is not yet implemented.
+ * Currently the legacy behavior is still implemented, where loopStart and loopEnd
+ * are in wrapping (overflow) frame units like the return value of getPosition().
+ * The plan is to fix all callers to use the new version at same time implementation changes.
+ *
* Parameters:
*
- * loopStart: loop start expressed as the number of PCM frames played since AudioTrack start.
- * loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start.
+ * loopStart: loop start in frames relative to start of buffer.
+ * loopEnd: loop end in frames relative to start of buffer.
* loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
- * pending or active loop. loopCount = -1 means infinite looping.
+ * pending or active loop. loopCount == -1 means infinite looping.
*
* For proper operation the following condition must be respected:
- * (loopEnd-loopStart) <= framecount()
+ * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount().
+ *
+ * If the loop period (loopEnd - loopStart) is too small for the implementation to support,
+ * setLoop() will return BAD_VALUE. loopCount must be >= -1.
+ *
*/
status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
/* Sets marker position. When playback reaches the number of frames specified, a callback with
* event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
* notification callback. To set a marker at a position which would compute as 0,
- * a workaround is to the set the marker at a nearby position such as -1 or 1.
+ * a workaround is to the set the marker at a nearby position such as ~0 or 1.
* If the AudioTrack has been opened with no callback function associated, the operation will
* fail.
*
@@ -354,18 +383,19 @@
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
- /* Sets playback head position within AudioTrack buffer. The new position is specified
- * in number of frames.
- * This method must be called with the AudioTrack in paused or stopped state.
- * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
- * Therefore using this method makes sense only when playing a "static" audio buffer
- * as opposed to streaming.
- * The getPosition() method on the other hand returns the total number of frames played since
- * playback start.
+ /* Sets playback head position.
+ * Only supported for static buffer mode.
+ *
+ * FIXME The comments below are for the new planned interpretation which is not yet implemented.
+ * Currently the legacy behavior is still implemented, where the new position
+ * is in wrapping (overflow) frame units like the return value of getPosition().
+ * The plan is to fix all callers to use the new version at same time implementation changes.
*
* Parameters:
*
- * position: New playback head position within AudioTrack buffer.
+ * position: New playback head position in frames relative to start of buffer.
+ * 0 <= position <= frameCount(). Note that end of buffer is permitted,
+ * but will result in an immediate underrun if started.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
@@ -378,8 +408,22 @@
/* Return the total number of frames played since playback start.
* The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz.
* It is reset to zero by flush(), reload(), and stop().
+ *
+ * Parameters:
+ *
+ * position: Address where to return play head position.
+ *
+ * Returned status (from utils/Errors.h) can be:
+ * - NO_ERROR: successful operation
+ * - BAD_VALUE: position is NULL
*/
- status_t getPosition(uint32_t *position);
+ status_t getPosition(uint32_t *position) const;
+
+ /* For static buffer mode only, this returns the current playback position in frames
+ * relative to start of buffer. It is analogous to the new API for
+ * setLoop() and setPosition(). After underrun, the position will be at end of buffer.
+ */
+ status_t getBufferPosition(uint32_t *position);
/* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
* rewriting the buffer before restarting playback after a stop.
@@ -426,15 +470,19 @@
*/
status_t attachAuxEffect(int effectId);
- /* Obtains a buffer of "frameCount" frames. The buffer must be
- * filled entirely, and then released with releaseBuffer().
- * If the track is stopped, obtainBuffer() returns
- * STOPPED instead of NO_ERROR as long as there are buffers available,
- * at which point NO_MORE_BUFFERS is returned.
+ /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
+ * After filling these slots with data, the caller should release them with releaseBuffer().
+ * If the track buffer is not full, obtainBuffer() returns as many contiguous
+ * [empty slots for] frames as are available immediately.
+ * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK
+ * regardless of the value of waitCount.
+ * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a
+ * maximum timeout based on waitCount; see chart below.
* Buffers will be returned until the pool
* is exhausted, at which point obtainBuffer() will either block
- * or return WOULD_BLOCK depending on the value of the "blocking"
+ * or return WOULD_BLOCK depending on the value of the "waitCount"
* parameter.
+ * Each sample is 16-bit signed PCM.
*
* obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
* which should use write() or callback EVENT_MORE_DATA instead.
@@ -457,24 +505,35 @@
* raw pointer to the buffer
*/
- enum {
- NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
- STOPPED = 1
- };
+ /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
+ status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+ __attribute__((__deprecated__));
- status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+private:
+ /* New internal API
+ * If nonContig is non-NULL, it is an output parameter that will be set to the number of
+ * additional non-contiguous frames that are available immediately.
+ * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(),
+ * in case the requested amount of frames is in two or more non-contiguous regions.
+ * FIXME requested and elapsed are both relative times. Consider changing to absolute time.
+ */
+ status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
+ struct timespec *elapsed = NULL, size_t *nonContig = NULL);
+public:
- /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
+ /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
+ // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
void releaseBuffer(Buffer* audioBuffer);
/* As a convenience we provide a write() interface to the audio buffer.
+ * Input parameter 'size' is in byte units.
* This is implemented on top of obtainBuffer/releaseBuffer. For best
* performance use callbacks. Returns actual number of bytes written >= 0,
* or one of the following negative status codes:
* INVALID_OPERATION AudioTrack is configured for shared buffer mode
* BAD_VALUE size is invalid
- * STOPPED AudioTrack was stopped during the write
- * NO_MORE_BUFFERS when obtainBuffer() returns same
+ * WOULD_BLOCK when obtainBuffer() returns same, or
+ * AudioTrack was stopped during the write
* or any other error code returned by IAudioTrack::start() or restoreTrack_l().
* Not supported for static buffer mode.
*/
@@ -483,7 +542,13 @@
/*
* Dumps the state of an audio track.
*/
- status_t dump(int fd, const Vector<String16>& args) const;
+ status_t dump(int fd, const Vector<String16>& args) const;
+
+ /*
+ * Return the total number of frames which AudioFlinger desired but were unavailable,
+ * and thus which resulted in an underrun. Reset to zero by stop().
+ */
+ uint32_t getUnderrunFrames() const;
protected:
/* copying audio tracks is not allowed */
@@ -502,19 +567,29 @@
void pause(); // suspend thread from execution at next loop boundary
void resume(); // allow thread to execute, if not requested to exit
+ void pauseConditional();
+ // like pause(), but only if prior resume() wasn't latched
private:
friend class AudioTrack;
virtual bool threadLoop();
- AudioTrack& mReceiver;
- ~AudioTrackThread();
+ AudioTrack& mReceiver;
+ virtual ~AudioTrackThread();
Mutex mMyLock; // Thread::mLock is private
Condition mMyCond; // Thread::mThreadExitedCondition is private
bool mPaused; // whether thread is currently paused
+ bool mResumeLatch; // whether next pauseConditional() will be a nop
};
// body of AudioTrackThread::threadLoop()
- bool processAudioBuffer(const sp<AudioTrackThread>& thread);
+ // returns the maximum amount of time before we would like to run again, where:
+ // 0 immediately
+ // > 0 no later than this many nanoseconds from now
+ // NS_WHENEVER still active but no particular deadline
+ // NS_INACTIVE inactive so don't run again until re-started
+ // NS_NEVER never again
+ static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
+ nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
// caller must hold lock on mLock for all _l methods
status_t createTrack_l(audio_stream_type_t streamType,
@@ -523,20 +598,24 @@
size_t frameCount,
audio_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output);
+ audio_io_handle_t output,
+ size_t epoch);
- // can only be called when !mActive
+ // can only be called when mState != STATE_ACTIVE
void flush_l();
- status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
+ void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
audio_io_handle_t getOutput_l();
- status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
- bool stopped_l() const { return !mActive; }
+ // FIXME enum is faster than strcmp() for parameter 'from'
+ status_t restoreTrack_l(const char *from);
+
+ // may be changed if IAudioTrack is re-created
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
- sp<AudioTrackThread> mAudioTrackThread;
+ audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ sp<AudioTrackThread> mAudioTrackThread;
float mVolume[2];
float mSendLevel;
uint32_t mSampleRate;
@@ -544,62 +623,89 @@
size_t mReqFrameCount; // frame count to request the next time a new
// IAudioTrack is needed
- audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
- // Starting address of buffers in shared memory. If there is a shared buffer, mBuffers
- // is the value of pointer() for the shared buffer, otherwise mBuffers points
- // immediately after the control block. This address is for the mapping within client
- // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
- void* mBuffers;
-
+ // constant after constructor or set()
audio_format_t mFormat; // as requested by client, not forced to 16-bit
audio_stream_type_t mStreamType;
uint32_t mChannelCount;
audio_channel_mask_t mChannelMask;
+ transfer_type mTransfer;
- // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.
- // For 8-bit PCM data, mFrameSizeAF is
- // twice as large because data is expanded to 16-bit before being stored in buffer.
+ // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's
+ // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
size_t mFrameSize; // app-level frame size
size_t mFrameSizeAF; // AudioFlinger frame size
status_t mStatus;
- uint32_t mLatency;
- bool mActive; // protected by mLock
+ // can change dynamically when IAudioTrack invalidated
+ uint32_t mLatency; // in ms
+
+ // Indicates the current track state. Protected by mLock.
+ enum State {
+ STATE_ACTIVE,
+ STATE_STOPPED,
+ STATE_PAUSED,
+ STATE_FLUSHED,
+ } mState;
callback_t mCbf; // callback handler for events, or NULL
void* mUserData; // for client callback handler
// for notification APIs
uint32_t mNotificationFramesReq; // requested number of frames between each
- // notification callback
+ // notification callback,
+ // at initial source sample rate
uint32_t mNotificationFramesAct; // actual number of frames between each
- // notification callback
+ // notification callback,
+ // at initial source sample rate
+ bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+
+ // These are private to processAudioBuffer(), and are not protected by a lock
+ uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
+ bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
+ int mObservedSequence; // last observed value of mSequence
+
sp<IMemory> mSharedBuffer;
- int mLoopCount;
- uint32_t mRemainingFrames;
+ uint32_t mLoopPeriod; // in frames, zero means looping is disabled
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
uint32_t mNewPosition; // in frames
- uint32_t mUpdatePeriod; // in frames
+ uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
- bool mFlushed; // FIXME will be made obsolete by making flush() synchronous
audio_output_flags_t mFlags;
int mSessionId;
int mAuxEffectId;
- // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock:
- // 1. mLock
- // 2. mCblk->lock
- // It is OK to lock only mCblk->lock.
mutable Mutex mLock;
bool mIsTimed;
int mPreviousPriority; // before start()
SchedPolicy mPreviousSchedulingGroup;
- AudioTrackClientProxy* mProxy;
bool mAwaitBoost; // thread should wait for priority boost before running
+
+ // The proxy should only be referenced while a lock is held because the proxy isn't
+ // multi-thread safe, especially the SingleStateQueue part of the proxy.
+ // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock,
+ // provided that the caller also holds an extra reference to the proxy and shared memory to keep
+ // them around in case they are replaced during the obtainBuffer().
+ sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only
+ sp<AudioTrackClientProxy> mProxy; // primary owner of the memory
+
+ bool mInUnderrun; // whether track is currently in underrun state
+
+private:
+ class DeathNotifier : public IBinder::DeathRecipient {
+ public:
+ DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { }
+ protected:
+ virtual void binderDied(const wp<IBinder>& who);
+ private:
+ const wp<AudioTrack> mAudioTrack;
+ };
+
+ sp<DeathNotifier> mDeathNotifier;
+ uint32_t mSequence; // incremented for each new IAudioTrack attempt
};
class TimedAudioTrack : public AudioTrack
diff --git a/include/media/IHDCP.h b/include/media/IHDCP.h
index 6d27b18..54fefa3 100644
--- a/include/media/IHDCP.h
+++ b/include/media/IHDCP.h
@@ -17,6 +17,7 @@
#include <binder/IInterface.h>
#include <media/hardware/HDCPAPI.h>
#include <media/stagefright/foundation/ABase.h>
+#include <ui/GraphicBuffer.h>
namespace android {
@@ -59,6 +60,20 @@
const void *inData, size_t size, uint32_t streamCTR,
uint64_t *outInputCTR, void *outData) = 0;
+ // Encrypt data according to the HDCP spec. "size" bytes of data starting
+ // at location "offset" are available in "buffer" (buffer handle). "size"
+ // may not be a multiple of 128 bits (16 bytes). An equal number of
+ // encrypted bytes should be written to the buffer at "outData" (virtual
+ // address). This operation is to be synchronous, i.e. this call does not
+ // return until outData contains size bytes of encrypted data.
+ // streamCTR will be assigned by the caller (to 0 for the first PES stream,
+ // 1 for the second and so on)
+ // inputCTR _will_be_maintained_by_the_callee_ for each PES stream.
+ virtual status_t encryptNative(
+ const sp<GraphicBuffer> &graphicBuffer,
+ size_t offset, size_t size, uint32_t streamCTR,
+ uint64_t *outInputCTR, void *outData) = 0;
+
// DECRYPTION only:
// Decrypt data according to the HDCP spec.
// "size" bytes of encrypted data are available at "inData"
diff --git a/include/media/JetPlayer.h b/include/media/JetPlayer.h
index 0616bf0..388f767 100644
--- a/include/media/JetPlayer.h
+++ b/include/media/JetPlayer.h
@@ -88,7 +88,7 @@
EAS_DATA_HANDLE mEasData;
EAS_FILE_LOCATOR mEasJetFileLoc;
EAS_PCM* mAudioBuffer;// EAS renders the MIDI data into this buffer,
- AudioTrack* mAudioTrack; // and we play it in this audio track
+ sp<AudioTrack> mAudioTrack; // and we play it in this audio track
int mTrackBufferSize;
S_JET_STATUS mJetStatus;
S_JET_STATUS mPreviousJetStatus;
diff --git a/include/media/SoundPool.h b/include/media/SoundPool.h
index 7bf3069..9e5654f 100644
--- a/include/media/SoundPool.h
+++ b/include/media/SoundPool.h
@@ -118,7 +118,7 @@
class SoundChannel : public SoundEvent {
public:
enum state { IDLE, RESUMING, STOPPING, PAUSED, PLAYING };
- SoundChannel() : mAudioTrack(NULL), mState(IDLE), mNumChannels(1),
+ SoundChannel() : mState(IDLE), mNumChannels(1),
mPos(0), mToggle(0), mAutoPaused(false) {}
~SoundChannel();
void init(SoundPool* soundPool);
@@ -148,7 +148,7 @@
bool doStop_l();
SoundPool* mSoundPool;
- AudioTrack* mAudioTrack;
+ sp<AudioTrack> mAudioTrack;
SoundEvent mNextEvent;
Mutex mLock;
int mState;
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 2183fbe..98c4332 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -160,7 +160,7 @@
bool isInited() { return (mState == TONE_IDLE)?false:true;}
// returns the audio session this ToneGenerator belongs to or 0 if an error occured.
- int getSessionId() { return (mpAudioTrack == NULL) ? 0 : mpAudioTrack->getSessionId(); }
+ int getSessionId() { return (mpAudioTrack == 0) ? 0 : mpAudioTrack->getSessionId(); }
private:
@@ -264,7 +264,7 @@
unsigned short mLoopCounter; // Current tone loopback count
uint32_t mSamplingRate; // AudioFlinger Sampling rate
- AudioTrack *mpAudioTrack; // Pointer to audio track used for playback
+ sp<AudioTrack> mpAudioTrack; // Pointer to audio track used for playback
Mutex mLock; // Mutex to control concurent access to ToneGenerator object from audio callback and application API
Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond
Condition mWaitCbkCond; // condition enabling interface to wait for audio callback completion after a change is requested
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index df25d7b..8876c9b 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -182,7 +182,7 @@
bool mSentFormat;
bool mIsEncoder;
-
+ bool mUseMetadataOnEncoderOutput;
bool mShutdownInProgress;
// If "mKeepComponentAllocated" we only transition back to Loaded state
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 1dc408f..3bf046d 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -70,7 +70,7 @@
private:
friend class VideoEditorAudioPlayer;
sp<MediaSource> mSource;
- AudioTrack *mAudioTrack;
+ sp<AudioTrack> mAudioTrack;
MediaBuffer *mInputBuffer;
diff --git a/include/media/stagefright/AudioSource.h b/include/media/stagefright/AudioSource.h
index 99f3c3b..4c9aaad 100644
--- a/include/media/stagefright/AudioSource.h
+++ b/include/media/stagefright/AudioSource.h
@@ -73,7 +73,7 @@
Condition mFrameAvailableCondition;
Condition mFrameEncodingCompletionCondition;
- AudioRecord *mRecord;
+ sp<AudioRecord> mRecord;
status_t mInitCheck;
bool mStarted;
int32_t mSampleRate;
diff --git a/include/media/stagefright/SurfaceMediaSource.h b/include/media/stagefright/SurfaceMediaSource.h
index 5f21da9..7d40379 100644
--- a/include/media/stagefright/SurfaceMediaSource.h
+++ b/include/media/stagefright/SurfaceMediaSource.h
@@ -146,9 +146,13 @@
// this consumer
sp<BufferQueue> mBufferQueue;
- // mBufferSlot caches GraphicBuffers from the buffer queue
- sp<GraphicBuffer> mBufferSlot[BufferQueue::NUM_BUFFER_SLOTS];
+ struct SlotData {
+ sp<GraphicBuffer> mGraphicBuffer;
+ uint64_t mFrameNumber;
+ };
+ // mSlots caches GraphicBuffers and frameNumbers from the buffer queue
+ SlotData mSlots[BufferQueue::NUM_BUFFER_SLOTS];
// The permenent width and height of SMS buffers
int mWidth;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 41e20f8..681f557 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -22,32 +22,46 @@
#include <utils/threads.h>
#include <utils/Log.h>
+#include <utils/RefBase.h>
+#include <media/nbaio/roundup.h>
+#include <media/SingleStateQueue.h>
+#include <private/media/StaticAudioTrackState.h>
namespace android {
// ----------------------------------------------------------------------------
-// Maximum cumulated timeout milliseconds before restarting audioflinger thread
-#define MAX_STARTUP_TIMEOUT_MS 3000 // Longer timeout period at startup to cope with A2DP
- // init time
-#define MAX_RUN_TIMEOUT_MS 1000
-#define WAIT_PERIOD_MS 10
-
-#define CBLK_UNDERRUN 0x01 // set: underrun (out) or overrrun (in), clear: no underrun or overrun
+#define CBLK_UNDERRUN 0x01 // set by server immediately on output underrun, cleared by client
#define CBLK_FORCEREADY 0x02 // set: track is considered ready immediately by AudioFlinger,
// clear: track is ready when buffer full
#define CBLK_INVALID 0x04 // track buffer invalidated by AudioFlinger, need to re-create
-#define CBLK_DISABLED 0x08 // track disabled by AudioFlinger due to underrun, need to re-start
+#define CBLK_DISABLED 0x08 // output track disabled by AudioFlinger due to underrun,
+ // need to re-start. Unlike CBLK_UNDERRUN, this is not set
+ // immediately, but only after a long string of underruns.
+// 0x10 unused
+#define CBLK_LOOP_CYCLE 0x20 // set by server each time a loop cycle other than final one completes
+#define CBLK_LOOP_FINAL 0x40 // set by server when the final loop cycle completes
+#define CBLK_BUFFER_END 0x80 // set by server when the position reaches end of buffer if not looping
+#define CBLK_OVERRUN 0x100 // set by server immediately on input overrun, cleared by client
+#define CBLK_INTERRUPT 0x200 // set by client on interrupt(), cleared by client in obtainBuffer()
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
- volatile int32_t mFront;
- volatile int32_t mRear;
+ // in continuously incrementing frame units, take modulo buffer size, which must be a power of 2
+ volatile int32_t mFront; // read by server
+ volatile int32_t mRear; // write by client
+ volatile int32_t mFlush; // incremented by client to indicate a request to flush;
+ // server notices and discards all data between mFront and mRear
+ volatile uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame
};
-// future
+typedef SingleStateQueue<StaticAudioTrackState> StaticAudioTrackSingleStateQueue;
+
struct AudioTrackSharedStatic {
- int mReserved;
+ StaticAudioTrackSingleStateQueue::Shared
+ mSingleStateQueue;
+ size_t mBufferPosition; // updated asynchronously by server,
+ // "for entertainment purposes only"
};
// ----------------------------------------------------------------------------
@@ -55,65 +69,61 @@
// Important: do not add any virtual methods, including ~
struct audio_track_cblk_t
{
+ // Since the control block is always located in shared memory, this constructor
+ // is only used for placement new(). It is never used for regular new() or stack.
+ audio_track_cblk_t();
+ /*virtual*/ ~audio_track_cblk_t() { }
+
friend class Proxy;
+ friend class ClientProxy;
friend class AudioTrackClientProxy;
friend class AudioRecordClientProxy;
friend class ServerProxy;
+ friend class AudioTrackServerProxy;
+ friend class AudioRecordServerProxy;
// The data members are grouped so that members accessed frequently and in the same context
// are in the same line of data cache.
- Mutex lock; // sizeof(int)
- Condition cv; // sizeof(int)
- // next 4 are offsets within "buffers"
- volatile uint32_t user;
- volatile uint32_t server;
- uint32_t userBase;
- uint32_t serverBase;
-
- int mPad1; // unused, but preserves cache line alignment
+ volatile uint32_t server; // updated asynchronously by server,
+ // "for entertainment purposes only"
size_t frameCount_; // used during creation to pass actual track buffer size
// from AudioFlinger to client, and not referenced again
- // FIXME remove here and replace by createTrack() in/out parameter
+ // FIXME remove here and replace by createTrack() in/out
+ // parameter
// renamed to "_" to detect incorrect use
- // Cache line boundary (32 bytes)
+ volatile int32_t mFutex; // semaphore: down (P) by client,
+ // up (V) by server or binderDied() or interrupt()
- uint32_t loopStart;
- uint32_t loopEnd; // read-only for server, read/write for client
- int loopCount; // read/write for client
+private:
+
+ size_t mMinimum; // server wakes up client if available >= mMinimum
// Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
// Left channel is in [0:15], right channel is in [16:31].
// Always read and write the combined pair atomically.
// For AudioTrack only, not used by AudioRecord.
-private:
uint32_t mVolumeLR;
uint32_t mSampleRate; // AudioTrack only: client's requested sample rate in Hz
// or 0 == default. Write-only client, read-only server.
+ // client write-only, server read-only
+ uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0
+
uint8_t mPad2; // unused
public:
// read-only for client, server writes once at initialization and is then read-only
uint8_t mName; // normal tracks: track name, fast tracks: track index
- // used by client only
- uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting
- // audioflinger
-
- uint16_t waitTimeMs; // Cumulated wait time, used by client only
-private:
- // client write-only, server read-only
- uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0
-public:
volatile int32_t flags;
// Cache line boundary (32 bytes)
-#if 0
+public:
union {
AudioTrackSharedStreaming mStreaming;
AudioTrackSharedStatic mStatic;
@@ -121,25 +131,6 @@
} u;
// Cache line boundary (32 bytes)
-#endif
-
- // Since the control block is always located in shared memory, this constructor
- // is only used for placement new(). It is never used for regular new() or stack.
- audio_track_cblk_t();
-
-private:
- // if there is a shared buffer, "buffers" is the value of pointer() for the shared
- // buffer, otherwise "buffers" points immediately after the control block
- void* buffer(void *buffers, uint32_t frameSize, size_t offset) const;
-
- bool tryLock();
-
- // isOut == true means AudioTrack, isOut == false means AudioRecord
- bool stepServer(size_t stepCount, size_t frameCount, bool isOut);
- uint32_t stepUser(size_t stepCount, size_t frameCount, bool isOut);
- uint32_t framesAvailable(size_t frameCount, bool isOut);
- uint32_t framesAvailable_l(size_t frameCount, bool isOut);
- uint32_t framesReady(bool isOut);
};
// ----------------------------------------------------------------------------
@@ -147,29 +138,31 @@
// Proxy for shared memory control block, to isolate callers from needing to know the details.
// There is exactly one ClientProxy and one ServerProxy per shared memory control block.
// The proxies are located in normal memory, and are not multi-thread safe within a given side.
-class Proxy {
+class Proxy : public RefBase {
protected:
- Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
- : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize) { }
+ Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut,
+ bool clientInServer);
virtual ~Proxy() { }
public:
- void* buffer(size_t offset) const {
- return mCblk->buffer(mBuffers, mFrameSize, offset);
- }
+ struct Buffer {
+ size_t mFrameCount; // number of frames available in this buffer
+ void* mRaw; // pointer to first frame
+ size_t mNonContig; // number of additional non-contiguous frames available
+ };
protected:
// These refer to shared memory, and are virtual addresses with respect to the current process.
// They may have different virtual addresses within the other process.
- audio_track_cblk_t* const mCblk; // the control block
- void* const mBuffers; // starting address of buffers
+ audio_track_cblk_t* const mCblk; // the control block
+ void* const mBuffers; // starting address of buffers
- const size_t mFrameCount; // not necessarily a power of 2
- const size_t mFrameSize; // in bytes
-#if 0
- const size_t mFrameCountP2; // mFrameCount rounded to power of 2, streaming mode
-#endif
-
+ const size_t mFrameCount; // not necessarily a power of 2
+ const size_t mFrameSize; // in bytes
+ const size_t mFrameCountP2; // mFrameCount rounded to power of 2, streaming mode
+ const bool mIsOut; // true for AudioTrack, false for AudioRecord
+ const bool mClientInServer; // true for OutputTrack, false for AudioTrack & AudioRecord
+ bool mIsShutdown; // latch set to true when shared memory corruption detected
};
// ----------------------------------------------------------------------------
@@ -177,9 +170,86 @@
// Proxy seen by AudioTrack client and AudioRecord client
class ClientProxy : public Proxy {
protected:
- ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
- : Proxy(cblk, buffers, frameCount, frameSize) { }
+ ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
+ bool isOut, bool clientInServer);
virtual ~ClientProxy() { }
+
+public:
+ static const struct timespec kForever;
+ static const struct timespec kNonBlocking;
+
+ // Obtain a buffer with filled frames (reading) or empty frames (writing).
+ // It is permitted to call obtainBuffer() multiple times in succession, without any intervening
+ // calls to releaseBuffer(). In that case, the final obtainBuffer() is the one that effectively
+ // sets or extends the unreleased frame count.
+ // On entry:
+ // buffer->mFrameCount should be initialized to maximum number of desired frames,
+ // which must be > 0.
+ // buffer->mNonContig is unused.
+ // buffer->mRaw is unused.
+ // requested is the requested timeout in local monotonic delta time units:
+ // NULL or &kNonBlocking means non-blocking (zero timeout).
+ // &kForever means block forever (infinite timeout).
+ // Other values mean a specific timeout in local monotonic delta time units.
+ // elapsed is a pointer to a location that will hold the total local monotonic time that
+ // elapsed while blocked, or NULL if not needed.
+ // On exit:
+ // buffer->mFrameCount has the actual number of contiguous available frames,
+ // which is always 0 when the return status != NO_ERROR.
+ // buffer->mNonContig is the number of additional non-contiguous available frames.
+ // buffer->mRaw is a pointer to the first available frame,
+ // or NULL when buffer->mFrameCount == 0.
+ // The return status is one of:
+ // NO_ERROR Success, buffer->mFrameCount > 0.
+ // WOULD_BLOCK Non-blocking mode and no frames are available.
+ // TIMED_OUT Timeout occurred before any frames became available.
+ // This can happen even for infinite timeout, due to a spurious wakeup.
+ // In this case, the caller should investigate and then re-try as appropriate.
+ // DEAD_OBJECT Server has died or invalidated, caller should destroy this proxy and re-create.
+ // -EINTR Call has been interrupted. Look around to see why, and then perhaps try again.
+ // NO_INIT Shared memory is corrupt.
+ // BAD_VALUE On entry buffer == NULL or buffer->mFrameCount == 0.
+ status_t obtainBuffer(Buffer* buffer, const struct timespec *requested = NULL,
+ struct timespec *elapsed = NULL);
+
+ // Release (some of) the frames last obtained.
+ // On entry, buffer->mFrameCount should have the number of frames to release,
+ // which must (cumulatively) be <= the number of frames last obtained but not yet released.
+ // buffer->mRaw is ignored, but is normally same pointer returned by last obtainBuffer().
+ // It is permitted to call releaseBuffer() multiple times to release the frames in chunks.
+ // On exit:
+ // buffer->mFrameCount is zero.
+ // buffer->mRaw is NULL.
+ void releaseBuffer(Buffer* buffer);
+
+ // Call after detecting server's death
+ void binderDied();
+
+ // Call to force an obtainBuffer() to return quickly with -EINTR
+ void interrupt();
+
+ size_t getPosition() {
+ return mEpoch + mCblk->server;
+ }
+
+ void setEpoch(size_t epoch) {
+ mEpoch = epoch;
+ }
+
+ void setMinimum(size_t minimum) {
+ mCblk->mMinimum = minimum;
+ }
+
+ // Return the number of frames that would need to be obtained and released
+ // in order for the client to be aligned at start of buffer
+ virtual size_t getMisalignment();
+
+ size_t getEpoch() const {
+ return mEpoch;
+ }
+
+private:
+ size_t mEpoch;
};
// ----------------------------------------------------------------------------
@@ -187,8 +257,10 @@
// Proxy used by AudioTrack client, which also includes AudioFlinger::PlaybackThread::OutputTrack
class AudioTrackClientProxy : public ClientProxy {
public:
- AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
- : ClientProxy(cblk, buffers, frameCount, frameSize) { }
+ AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize, bool clientInServer = false)
+ : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/,
+ clientInServer) { }
virtual ~AudioTrackClientProxy() { }
// No barriers on the following operations, so the ordering of loads/stores
@@ -208,27 +280,36 @@
mCblk->mSampleRate = sampleRate;
}
- // called by:
- // PlaybackThread::OutputTrack::write
- // AudioTrack::createTrack_l
- // AudioTrack::releaseBuffer
- // AudioTrack::reload
- // AudioTrack::restoreTrack_l (2 places)
- size_t stepUser(size_t stepCount) {
- return mCblk->stepUser(stepCount, mFrameCount, true /*isOut*/);
+ virtual void flush();
+
+ virtual uint32_t getUnderrunFrames() const {
+ return mCblk->u.mStreaming.mUnderrunFrames;
+ }
+};
+
+class StaticAudioTrackClientProxy : public AudioTrackClientProxy {
+public:
+ StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize);
+ virtual ~StaticAudioTrackClientProxy() { }
+
+ virtual void flush();
+
+#define MIN_LOOP 16 // minimum length of each loop iteration in frames
+ void setLoop(size_t loopStart, size_t loopEnd, int loopCount);
+ size_t getBufferPosition();
+
+ virtual size_t getMisalignment() {
+ return 0;
}
- // called by AudioTrack::obtainBuffer and AudioTrack::processBuffer
- size_t framesAvailable() {
- return mCblk->framesAvailable(mFrameCount, true /*isOut*/);
+ virtual uint32_t getUnderrunFrames() const {
+ return 0;
}
- // called by AudioTrack::obtainBuffer and PlaybackThread::OutputTrack::obtainBuffer
- // FIXME remove this API since it assumes a lock that should be invisible to caller
- size_t framesAvailable_l() {
- return mCblk->framesAvailable_l(mFrameCount, true /*isOut*/);
- }
-
+private:
+ StaticAudioTrackSingleStateQueue::Mutator mMutator;
+ size_t mBufferPosition; // so that getBufferPosition() appears to be synchronous
};
// ----------------------------------------------------------------------------
@@ -236,60 +317,122 @@
// Proxy used by AudioRecord client
class AudioRecordClientProxy : public ClientProxy {
public:
- AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize)
- : ClientProxy(cblk, buffers, frameCount, frameSize) { }
+ AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize)
+ : ClientProxy(cblk, buffers, frameCount, frameSize,
+ false /*isOut*/, false /*clientInServer*/) { }
~AudioRecordClientProxy() { }
-
- // called by AudioRecord::releaseBuffer
- size_t stepUser(size_t stepCount) {
- return mCblk->stepUser(stepCount, mFrameCount, false /*isOut*/);
- }
-
- // called by AudioRecord::processBuffer
- size_t framesAvailable() {
- return mCblk->framesAvailable(mFrameCount, false /*isOut*/);
- }
-
- // called by AudioRecord::obtainBuffer
- size_t framesReady() {
- return mCblk->framesReady(false /*isOut*/);
- }
-
};
// ----------------------------------------------------------------------------
// Proxy used by AudioFlinger server
class ServerProxy : public Proxy {
+protected:
+ ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
+ bool isOut, bool clientInServer);
public:
- ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut)
- : Proxy(cblk, buffers, frameCount, frameSize), mIsOut(isOut) { }
virtual ~ServerProxy() { }
- // for AudioTrack and AudioRecord
- bool step(size_t stepCount) { return mCblk->stepServer(stepCount, mFrameCount, mIsOut); }
+ // Obtain a buffer with filled frames (writing) or empty frames (reading).
+ // It is permitted to call obtainBuffer() multiple times in succession, without any intervening
+ // calls to releaseBuffer(). In that case, the final obtainBuffer() is the one that effectively
+ // sets or extends the unreleased frame count.
+ // Always non-blocking.
+ // On entry:
+ // buffer->mFrameCount should be initialized to maximum number of desired frames,
+ // which must be > 0.
+ // buffer->mNonContig is unused.
+ // buffer->mRaw is unused.
+ // On exit:
+ // buffer->mFrameCount has the actual number of contiguous available frames,
+ // which is always 0 when the return status != NO_ERROR.
+ // buffer->mNonContig is the number of additional non-contiguous available frames.
+ // buffer->mRaw is a pointer to the first available frame,
+ // or NULL when buffer->mFrameCount == 0.
+ // The return status is one of:
+ // NO_ERROR Success, buffer->mFrameCount > 0.
+ // WOULD_BLOCK No frames are available.
+ // NO_INIT Shared memory is corrupt.
+ virtual status_t obtainBuffer(Buffer* buffer);
+ // Release (some of) the frames last obtained.
+ // On entry, buffer->mFrameCount should have the number of frames to release,
+ // which must (cumulatively) be <= the number of frames last obtained but not yet released.
+ // It is permitted to call releaseBuffer() multiple times to release the frames in chunks.
+ // buffer->mRaw is ignored, but is normally same pointer returned by last obtainBuffer().
+ // On exit:
+ // buffer->mFrameCount is zero.
+ // buffer->mRaw is NULL.
+ virtual void releaseBuffer(Buffer* buffer);
+
+protected:
+ size_t mUnreleased; // unreleased frames remaining from most recent obtainBuffer()
+ size_t mAvailToClient; // estimated frames available to client prior to releaseBuffer()
+private:
+ int32_t mFlush; // our copy of cblk->u.mStreaming.mFlush, for streaming output only
+ bool mDeferWake; // whether another releaseBuffer() is expected soon
+};
+
+// Proxy used by AudioFlinger for servicing AudioTrack
+class AudioTrackServerProxy : public ServerProxy {
+public:
+ AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize, bool clientInServer = false)
+ : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer) { }
+protected:
+ virtual ~AudioTrackServerProxy() { }
+
+public:
// return value of these methods must be validated by the caller
uint32_t getSampleRate() const { return mCblk->mSampleRate; }
uint16_t getSendLevel_U4_12() const { return mCblk->mSendLevel; }
uint32_t getVolumeLR() const { return mCblk->mVolumeLR; }
- // for AudioTrack only
- size_t framesReady() {
- ALOG_ASSERT(mIsOut);
- return mCblk->framesReady(true);
- }
+ // estimated total number of filled frames available to server to read,
+ // which may include non-contiguous frames
+ virtual size_t framesReady();
- // for AudioRecord only, called by RecordThread::RecordTrack::getNextBuffer
- // FIXME remove this API since it assumes a lock that should be invisible to caller
- size_t framesAvailableIn_l() {
- ALOG_ASSERT(!mIsOut);
- return mCblk->framesAvailable_l(mFrameCount, false);
- }
+ // Currently AudioFlinger will call framesReady() for a fast track from two threads:
+ // FastMixer thread, and normal mixer thread. This is dangerous, as the proxy is intended
+ // to be called from at most one thread of server, and one thread of client.
+ // As a temporary workaround, this method informs the proxy implementation that it
+ // should avoid doing a state queue poll from within framesReady().
+ // FIXME Change AudioFlinger to not call framesReady() from normal mixer thread.
+ virtual void framesReadyIsCalledByMultipleThreads() { }
+};
+
+class StaticAudioTrackServerProxy : public AudioTrackServerProxy {
+public:
+ StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize);
+protected:
+ virtual ~StaticAudioTrackServerProxy() { }
+
+public:
+ virtual size_t framesReady();
+ virtual void framesReadyIsCalledByMultipleThreads();
+ virtual status_t obtainBuffer(Buffer* buffer);
+ virtual void releaseBuffer(Buffer* buffer);
private:
- const bool mIsOut; // true for AudioTrack, false for AudioRecord
+ ssize_t pollPosition(); // poll for state queue update, and return current position
+ StaticAudioTrackSingleStateQueue::Observer mObserver;
+ size_t mPosition; // server's current play position in frames, relative to 0
+ size_t mEnd; // cached value computed from mState, safe for asynchronous read
+ bool mFramesReadyIsCalledByMultipleThreads;
+ StaticAudioTrackState mState;
+};
+// Proxy used by AudioFlinger for servicing AudioRecord
+class AudioRecordServerProxy : public ServerProxy {
+public:
+ AudioRecordServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize)
+ : ServerProxy(cblk, buffers, frameCount, frameSize, false /*isOut*/,
+ false /*clientInServer*/) { }
+protected:
+ virtual ~AudioRecordServerProxy() { }
};
// ----------------------------------------------------------------------------
diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
index c111ba8..3fa8b87 100755
--- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
@@ -35,8 +35,7 @@
VideoEditorAudioPlayer::VideoEditorAudioPlayer(
const sp<MediaPlayerBase::AudioSink> &audioSink,
PreviewPlayer *observer)
- : mAudioTrack(NULL),
- mInputBuffer(NULL),
+ : mInputBuffer(NULL),
mSampleRate(0),
mLatencyUs(0),
mFrameSize(0),
@@ -111,8 +110,7 @@
} else {
mAudioTrack->stop();
- delete mAudioTrack;
- mAudioTrack = NULL;
+ mAudioTrack.clear();
}
// Make sure to release any buffer we hold onto so that the
@@ -538,8 +536,7 @@
0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);
if ((err = mAudioTrack->initCheck()) != OK) {
- delete mAudioTrack;
- mAudioTrack = NULL;
+ mAudioTrack.clear();
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.h b/libvideoeditor/lvpp/VideoEditorAudioPlayer.h
index 626df39..a5616c1 100755
--- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.h
@@ -91,7 +91,7 @@
int64_t mBGAudioStoryBoardCurrentMediaVolumeVal;
sp<MediaSource> mSource;
- AudioTrack *mAudioTrack;
+ sp<AudioTrack> mAudioTrack;
MediaBuffer *mInputBuffer;
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 91a4415..4a14b40 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -310,7 +310,6 @@
VideoEditorPlayer::VeAudioOutput::VeAudioOutput()
: mCallback(NULL),
mCallbackCookie(NULL) {
- mTrack = 0;
mStreamType = AUDIO_STREAM_MUSIC;
mLeftVolume = 1.0;
mRightVolume = 1.0;
@@ -405,7 +404,7 @@
}
ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
- if (mTrack) close();
+ if (mTrack != 0) close();
uint32_t afSampleRate;
size_t afFrameCount;
int frameCount;
@@ -434,7 +433,7 @@
}
}
- AudioTrack *t;
+ sp<AudioTrack> t;
if (mCallback != NULL) {
t = new AudioTrack(
mStreamType,
@@ -457,7 +456,6 @@
if ((t == 0) || (t->initCheck() != NO_ERROR)) {
ALOGE("Unable to create audio track");
- delete t;
return NO_INIT;
}
@@ -472,7 +470,7 @@
void VideoEditorPlayer::VeAudioOutput::start() {
ALOGV("start");
- if (mTrack) {
+ if (mTrack != 0) {
mTrack->setVolume(mLeftVolume, mRightVolume);
mTrack->start();
mTrack->getPosition(&mNumFramesWritten);
@@ -492,7 +490,7 @@
LOG_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
//ALOGV("write(%p, %u)", buffer, size);
- if (mTrack) {
+ if (mTrack != 0) {
snoopWrite(buffer, size);
ssize_t ret = mTrack->write(buffer, size);
mNumFramesWritten += ret / 4; // assume 16 bit stereo
@@ -504,26 +502,25 @@
void VideoEditorPlayer::VeAudioOutput::stop() {
ALOGV("stop");
- if (mTrack) mTrack->stop();
+ if (mTrack != 0) mTrack->stop();
}
void VideoEditorPlayer::VeAudioOutput::flush() {
ALOGV("flush");
- if (mTrack) mTrack->flush();
+ if (mTrack != 0) mTrack->flush();
}
void VideoEditorPlayer::VeAudioOutput::pause() {
ALOGV("VeAudioOutput::pause");
- if (mTrack) mTrack->pause();
+ if (mTrack != 0) mTrack->pause();
}
void VideoEditorPlayer::VeAudioOutput::close() {
ALOGV("close");
- delete mTrack;
- mTrack = 0;
+ mTrack.clear();
}
void VideoEditorPlayer::VeAudioOutput::setVolume(float left, float right) {
@@ -531,7 +528,7 @@
ALOGV("setVolume(%f, %f)", left, right);
mLeftVolume = left;
mRightVolume = right;
- if (mTrack) {
+ if (mTrack != 0) {
mTrack->setVolume(left, right);
}
}
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index 77194ab..defc90d 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -71,7 +71,7 @@
static void CallbackWrapper(
int event, void *me, void *info);
- AudioTrack* mTrack;
+ sp<AudioTrack> mTrack;
AudioCallback mCallback;
void * mCallbackCookie;
audio_stream_type_t mStreamType;
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 2c0c3a5..96755bb 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -53,7 +53,8 @@
Visualizer.cpp \
MemoryLeakTrackUtil.cpp \
SoundPool.cpp \
- SoundPoolThread.cpp
+ SoundPoolThread.cpp \
+ StringArray.cpp
LOCAL_SRC_FILES += ../libnbaio/roundup.c
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 40ff1bf..9faa497 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -19,18 +19,13 @@
#define LOG_TAG "AudioRecord"
#include <sys/resource.h>
-#include <sys/types.h>
-
#include <binder/IPCThreadState.h>
-#include <cutils/atomic.h>
-#include <cutils/compiler.h>
#include <media/AudioRecord.h>
-#include <media/AudioSystem.h>
-#include <system/audio.h>
#include <utils/Log.h>
-
#include <private/media/AudioTrackShared.h>
+#define WAIT_PERIOD_MS 10
+
namespace android {
// ---------------------------------------------------------------------------
@@ -41,7 +36,9 @@
audio_format_t format,
audio_channel_mask_t channelMask)
{
- if (frameCount == NULL) return BAD_VALUE;
+ if (frameCount == NULL) {
+ return BAD_VALUE;
+ }
// default to 0 in case of error
*frameCount = 0;
@@ -75,8 +72,7 @@
AudioRecord::AudioRecord()
: mStatus(NO_INIT), mSessionId(0),
- mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),
- mProxy(NULL)
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
}
@@ -89,14 +85,15 @@
callback_t cbf,
void* user,
int notificationFrames,
- int sessionId)
+ int sessionId,
+ transfer_type transferType)
: mStatus(NO_INIT), mSessionId(0),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mProxy(NULL)
{
- mStatus = set(inputSource, sampleRate, format, channelMask,
- frameCount, cbf, user, notificationFrames, false /*threadCanCallJava*/, sessionId);
+ mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
+ notificationFrames, false /*threadCanCallJava*/, sessionId, transferType);
}
AudioRecord::~AudioRecord()
@@ -111,11 +108,13 @@
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
- mAudioRecord.clear();
+ if (mAudioRecord != 0) {
+ mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mAudioRecord.clear();
+ }
IPCThreadState::self()->flushCommands();
AudioSystem::releaseAudioSessionId(mSessionId);
}
- delete mProxy;
}
status_t AudioRecord::set(
@@ -128,8 +127,32 @@
void* user,
int notificationFrames,
bool threadCanCallJava,
- int sessionId)
+ int sessionId,
+ transfer_type transferType)
{
+ switch (transferType) {
+ case TRANSFER_DEFAULT:
+ if (cbf == NULL || threadCanCallJava) {
+ transferType = TRANSFER_SYNC;
+ } else {
+ transferType = TRANSFER_CALLBACK;
+ }
+ break;
+ case TRANSFER_CALLBACK:
+ if (cbf == NULL) {
+ ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
+ return BAD_VALUE;
+ }
+ break;
+ case TRANSFER_OBTAIN:
+ case TRANSFER_SYNC:
+ break;
+ default:
+ ALOGE("Invalid transfer type %d", transferType);
+ return BAD_VALUE;
+ }
+ mTransfer = transferType;
+
// FIXME "int" here is legacy and will be replaced by size_t later
if (frameCountInt < 0) {
ALOGE("Invalid frame count %d", frameCountInt);
@@ -143,6 +166,7 @@
AutoMutex lock(mLock);
if (mAudioRecord != 0) {
+ ALOGE("Track already in use");
return INVALID_OPERATION;
}
@@ -159,14 +183,16 @@
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
+
// validate parameters
if (!audio_is_valid_format(format)) {
- ALOGE("Invalid format");
+ ALOGE("Invalid format %d", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
+ ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
@@ -200,6 +226,7 @@
size_t minFrameCount = 0;
status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask);
if (status != NO_ERROR) {
+ ALOGE("getMinFrameCount() failed; status %d", status);
return status;
}
ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
@@ -207,6 +234,7 @@
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
+ ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
return BAD_VALUE;
}
@@ -215,7 +243,7 @@
}
// create the IAudioRecord
- status = openRecord_l(sampleRate, format, frameCount, input);
+ status = openRecord_l(sampleRate, format, frameCount, input, 0 /*epoch*/);
if (status != NO_ERROR) {
return status;
}
@@ -233,7 +261,7 @@
mActive = false;
mCbf = cbf;
mNotificationFrames = notificationFrames;
- mRemainingFrames = notificationFrames;
+ mRefreshRemaining = true;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
@@ -244,117 +272,78 @@
mInputSource = inputSource;
mInput = input;
AudioSystem::acquireAudioSessionId(mSessionId);
+ mSequence = 1;
+ mObservedSequence = mSequence;
+ mInOverrun = false;
return NO_ERROR;
}
-status_t AudioRecord::initCheck() const
-{
- return mStatus;
-}
-
-// -------------------------------------------------------------------------
-
-uint32_t AudioRecord::latency() const
-{
- return mLatency;
-}
-
-audio_format_t AudioRecord::format() const
-{
- return mFormat;
-}
-
-uint32_t AudioRecord::channelCount() const
-{
- return mChannelCount;
-}
-
-size_t AudioRecord::frameCount() const
-{
- return mFrameCount;
-}
-
-audio_source_t AudioRecord::inputSource() const
-{
- return mInputSource;
-}
-
// -------------------------------------------------------------------------
status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
{
- status_t ret = NO_ERROR;
- sp<AudioRecordThread> t = mAudioRecordThread;
-
ALOGV("start, sync event %d trigger session %d", event, triggerSession);
AutoMutex lock(mLock);
- // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
- // while we are accessing the cblk
- sp<IAudioRecord> audioRecord = mAudioRecord;
- sp<IMemory> iMem = mCblkMemory;
- audio_track_cblk_t* cblk = mCblk;
+ if (mActive) {
+ return NO_ERROR;
+ }
- if (!mActive) {
+ // reset current position as seen by client to 0
+ mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
+
+ mNewPosition = mProxy->getPosition() + mUpdatePeriod;
+ int32_t flags = android_atomic_acquire_load(&mCblk->flags);
+
+ status_t status = NO_ERROR;
+ if (!(flags & CBLK_INVALID)) {
+ ALOGV("mAudioRecord->start()");
+ status = mAudioRecord->start(event, triggerSession);
+ if (status == DEAD_OBJECT) {
+ flags |= CBLK_INVALID;
+ }
+ }
+ if (flags & CBLK_INVALID) {
+ status = restoreRecord_l("start");
+ }
+
+ if (status != NO_ERROR) {
+ ALOGE("start() status %d", status);
+ } else {
mActive = true;
-
- cblk->lock.lock();
- if (!(cblk->flags & CBLK_INVALID)) {
- cblk->lock.unlock();
- ALOGV("mAudioRecord->start()");
- ret = mAudioRecord->start(event, triggerSession);
- cblk->lock.lock();
- if (ret == DEAD_OBJECT) {
- android_atomic_or(CBLK_INVALID, &cblk->flags);
- }
- }
- if (cblk->flags & CBLK_INVALID) {
- audio_track_cblk_t* temp = cblk;
- ret = restoreRecord_l(temp);
- cblk = temp;
- }
- cblk->lock.unlock();
- if (ret == NO_ERROR) {
- mNewPosition = cblk->user + mUpdatePeriod;
- cblk->bufferTimeoutMs = (event == AudioSystem::SYNC_EVENT_NONE) ? MAX_RUN_TIMEOUT_MS :
- AudioSystem::kSyncRecordStartTimeOutMs;
- cblk->waitTimeMs = 0;
- if (t != 0) {
- t->resume();
- } else {
- mPreviousPriority = getpriority(PRIO_PROCESS, 0);
- get_sched_policy(0, &mPreviousSchedulingGroup);
- androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
- }
+ sp<AudioRecordThread> t = mAudioRecordThread;
+ if (t != 0) {
+ t->resume();
} else {
- mActive = false;
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ get_sched_policy(0, &mPreviousSchedulingGroup);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
}
- return ret;
+ return status;
}
void AudioRecord::stop()
{
- sp<AudioRecordThread> t = mAudioRecordThread;
-
- ALOGV("stop");
-
AutoMutex lock(mLock);
- if (mActive) {
- mActive = false;
- mCblk->cv.signal();
- mAudioRecord->stop();
- // the record head position will reset to 0, so if a marker is set, we need
- // to activate it again
- mMarkerReached = false;
- if (t != 0) {
- t->pause();
- } else {
- setpriority(PRIO_PROCESS, 0, mPreviousPriority);
- set_sched_policy(0, mPreviousSchedulingGroup);
- }
+ if (!mActive) {
+ return;
+ }
+
+ mActive = false;
+ mProxy->interrupt();
+ mAudioRecord->stop();
+ // the record head position will reset to 0, so if a marker is set, we need
+ // to activate it again
+ mMarkerReached = false;
+ sp<AudioRecordThread> t = mAudioRecordThread;
+ if (t != 0) {
+ t->pause();
+ } else {
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ set_sched_policy(0, mPreviousSchedulingGroup);
}
}
@@ -364,14 +353,11 @@
return !mActive;
}
-uint32_t AudioRecord::getSampleRate() const
-{
- return mSampleRate;
-}
-
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
- if (mCbf == NULL) return INVALID_OPERATION;
+ if (mCbf == NULL) {
+ return INVALID_OPERATION;
+ }
AutoMutex lock(mLock);
mMarkerPosition = marker;
@@ -382,7 +368,9 @@
status_t AudioRecord::getMarkerPosition(uint32_t *marker) const
{
- if (marker == NULL) return BAD_VALUE;
+ if (marker == NULL) {
+ return BAD_VALUE;
+ }
AutoMutex lock(mLock);
*marker = mMarkerPosition;
@@ -392,13 +380,12 @@
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == NULL) return INVALID_OPERATION;
-
- uint32_t curPosition;
- getPosition(&curPosition);
+ if (mCbf == NULL) {
+ return INVALID_OPERATION;
+ }
AutoMutex lock(mLock);
- mNewPosition = curPosition + updatePeriod;
+ mNewPosition = mProxy->getPosition() + updatePeriod;
mUpdatePeriod = updatePeriod;
return NO_ERROR;
@@ -406,7 +393,9 @@
status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
- if (updatePeriod == NULL) return BAD_VALUE;
+ if (updatePeriod == NULL) {
+ return BAD_VALUE;
+ }
AutoMutex lock(mLock);
*updatePeriod = mUpdatePeriod;
@@ -416,10 +405,12 @@
status_t AudioRecord::getPosition(uint32_t *position) const
{
- if (position == NULL) return BAD_VALUE;
+ if (position == NULL) {
+ return BAD_VALUE;
+ }
AutoMutex lock(mLock);
- *position = mCblk->user;
+ *position = mProxy->getPosition();
return NO_ERROR;
}
@@ -427,7 +418,7 @@
unsigned int AudioRecord::getInputFramesLost() const
{
// no need to check mActive, because if inactive this will return 0, which is what we want
- return AudioSystem::getInputFramesLost(mInput);
+ return AudioSystem::getInputFramesLost(getInput());
}
// -------------------------------------------------------------------------
@@ -437,7 +428,8 @@
uint32_t sampleRate,
audio_format_t format,
size_t frameCount,
- audio_io_handle_t input)
+ audio_io_handle_t input,
+ size_t epoch)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -447,7 +439,7 @@
}
pid_t tid = -1;
- // FIXME see similar logic at AudioTrack
+ // FIXME see similar logic at AudioTrack for tid
int originalSessionId = mSessionId;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
@@ -470,133 +462,138 @@
ALOGE("Could not get control block");
return NO_INIT;
}
- mAudioRecord.clear();
+ if (mAudioRecord != 0) {
+ mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mDeathNotifier.clear();
+ }
mAudioRecord = record;
- mCblkMemory.clear();
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
mCblk = cblk;
- mBuffers = (char*)cblk + sizeof(audio_track_cblk_t);
- cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
- cblk->waitTimeMs = 0;
+
+ // starting address of buffers in shared memory
+ void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
// update proxy
- delete mProxy;
- mProxy = new AudioRecordClientProxy(cblk, mBuffers, frameCount, mFrameSize);
+ mProxy = new AudioRecordClientProxy(cblk, buffers, frameCount, mFrameSize);
+ mProxy->setEpoch(epoch);
+ mProxy->setMinimum(mNotificationFrames);
+
+ mDeathNotifier = new DeathNotifier(this);
+ mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
- ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
+ if (audioBuffer == NULL) {
+ return BAD_VALUE;
+ }
+ if (mTransfer != TRANSFER_OBTAIN) {
+ audioBuffer->frameCount = 0;
+ audioBuffer->size = 0;
+ audioBuffer->raw = NULL;
+ return INVALID_OPERATION;
+ }
- AutoMutex lock(mLock);
- bool active;
- status_t result = NO_ERROR;
- audio_track_cblk_t* cblk = mCblk;
- uint32_t framesReq = audioBuffer->frameCount;
- uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
+ const struct timespec *requested;
+ if (waitCount == -1) {
+ requested = &ClientProxy::kForever;
+ } else if (waitCount == 0) {
+ requested = &ClientProxy::kNonBlocking;
+ } else if (waitCount > 0) {
+ long long ms = WAIT_PERIOD_MS * (long long) waitCount;
+ struct timespec timeout;
+ timeout.tv_sec = ms / 1000;
+ timeout.tv_nsec = (int) (ms % 1000) * 1000000;
+ requested = &timeout;
+ } else {
+ ALOGE("%s invalid waitCount %d", __func__, waitCount);
+ requested = NULL;
+ }
+ return obtainBuffer(audioBuffer, requested);
+}
- audioBuffer->frameCount = 0;
- audioBuffer->size = 0;
+status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
+ struct timespec *elapsed, size_t *nonContig)
+{
+ // previous and new IAudioRecord sequence numbers are used to detect track re-creation
+ uint32_t oldSequence = 0;
+ uint32_t newSequence;
- size_t framesReady = mProxy->framesReady();
+ Proxy::Buffer buffer;
+ status_t status = NO_ERROR;
- if (framesReady == 0) {
- cblk->lock.lock();
- goto start_loop_here;
- while (framesReady == 0) {
- active = mActive;
- if (CC_UNLIKELY(!active)) {
- cblk->lock.unlock();
- return NO_MORE_BUFFERS;
- }
- if (CC_UNLIKELY(!waitCount)) {
- cblk->lock.unlock();
- return WOULD_BLOCK;
- }
- if (!(cblk->flags & CBLK_INVALID)) {
- mLock.unlock();
- // this condition is in shared memory, so if IAudioRecord and control block
- // are replaced due to mediaserver death or IAudioRecord invalidation then
- // cv won't be signalled, but fortunately the timeout will limit the wait
- result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
- cblk->lock.unlock();
- mLock.lock();
- if (!mActive) {
- return status_t(STOPPED);
- }
- // IAudioRecord may have been re-created while mLock was unlocked
- cblk = mCblk;
- cblk->lock.lock();
- }
- if (cblk->flags & CBLK_INVALID) {
- goto create_new_record;
- }
- if (CC_UNLIKELY(result != NO_ERROR)) {
- cblk->waitTimeMs += waitTimeMs;
- if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
- ALOGW( "obtainBuffer timed out (is the CPU pegged?) "
- "user=%08x, server=%08x", cblk->user, cblk->server);
- cblk->lock.unlock();
- // callback thread or sync event hasn't changed
- result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
- cblk->lock.lock();
- if (result == DEAD_OBJECT) {
- android_atomic_or(CBLK_INVALID, &cblk->flags);
-create_new_record:
- audio_track_cblk_t* temp = cblk;
- result = AudioRecord::restoreRecord_l(temp);
- cblk = temp;
+ static const int32_t kMaxTries = 5;
+ int32_t tryCounter = kMaxTries;
+
+ do {
+ // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
+ // keep them from going away if another thread re-creates the track during obtainBuffer()
+ sp<AudioRecordClientProxy> proxy;
+ sp<IMemory> iMem;
+ {
+ // start of lock scope
+ AutoMutex lock(mLock);
+
+ newSequence = mSequence;
+ // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
+ if (status == DEAD_OBJECT) {
+ // re-create track, unless someone else has already done so
+ if (newSequence == oldSequence) {
+ status = restoreRecord_l("obtainBuffer");
+ if (status != NO_ERROR) {
+ break;
}
- if (result != NO_ERROR) {
- ALOGW("obtainBuffer create Track error %d", result);
- cblk->lock.unlock();
- return result;
- }
- cblk->waitTimeMs = 0;
- }
- if (--waitCount == 0) {
- cblk->lock.unlock();
- return TIMED_OUT;
}
}
- // read the server count again
- start_loop_here:
- framesReady = mProxy->framesReady();
- }
- cblk->lock.unlock();
+ oldSequence = newSequence;
+
+ // Keep the extra references
+ proxy = mProxy;
+ iMem = mCblkMemory;
+
+ // Non-blocking if track is stopped
+ if (!mActive) {
+ requested = &ClientProxy::kNonBlocking;
+ }
+
+ } // end of lock scope
+
+ buffer.mFrameCount = audioBuffer->frameCount;
+ // FIXME starts the requested timeout and elapsed over from scratch
+ status = proxy->obtainBuffer(&buffer, requested, elapsed);
+
+ } while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
+
+ audioBuffer->frameCount = buffer.mFrameCount;
+ audioBuffer->size = buffer.mFrameCount * mFrameSize;
+ audioBuffer->raw = buffer.mRaw;
+ if (nonContig != NULL) {
+ *nonContig = buffer.mNonContig;
}
-
- cblk->waitTimeMs = 0;
- // reset time out to running value after obtaining a buffer
- cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
-
- if (framesReq > framesReady) {
- framesReq = framesReady;
- }
-
- uint32_t u = cblk->user;
- uint32_t bufferEnd = cblk->userBase + mFrameCount;
-
- if (framesReq > bufferEnd - u) {
- framesReq = bufferEnd - u;
- }
-
- audioBuffer->frameCount = framesReq;
- audioBuffer->size = framesReq * mFrameSize;
- audioBuffer->raw = mProxy->buffer(u);
- active = mActive;
- return active ? status_t(NO_ERROR) : status_t(STOPPED);
+ return status;
}
void AudioRecord::releaseBuffer(Buffer* audioBuffer)
{
- ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
+ // all TRANSFER_* are valid
+
+ size_t stepCount = audioBuffer->size / mFrameSize;
+ if (stepCount == 0) {
+ return;
+ }
+
+ Proxy::Buffer buffer;
+ buffer.mFrameCount = stepCount;
+ buffer.mRaw = audioBuffer->raw;
AutoMutex lock(mLock);
- (void) mProxy->stepUser(audioBuffer->frameCount);
+ mInOverrun = false;
+ mProxy->releaseBuffer(&buffer);
+
+ // the server does not automatically disable recorder on overrun, so no need to restart
}
audio_io_handle_t AudioRecord::getInput() const
@@ -616,215 +613,304 @@
return mInput;
}
-int AudioRecord::getSessionId() const
-{
- // no lock needed because session ID doesn't change after first set()
- return mSessionId;
-}
-
// -------------------------------------------------------------------------
ssize_t AudioRecord::read(void* buffer, size_t userSize)
{
- ssize_t read = 0;
- Buffer audioBuffer;
- int8_t *dst = static_cast<int8_t*>(buffer);
+ if (mTransfer != TRANSFER_SYNC) {
+ return INVALID_OPERATION;
+ }
- if (ssize_t(userSize) < 0) {
- // sanity-check. user is most-likely passing an error code.
- ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)",
- buffer, userSize, userSize);
+ if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
+ // sanity-check. user is most-likely passing an error code, and it would
+ // make the return value ambiguous (actualSize vs error).
+ ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
return BAD_VALUE;
}
- mLock.lock();
- // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
- // while we are accessing the cblk
- sp<IAudioRecord> audioRecord = mAudioRecord;
- sp<IMemory> iMem = mCblkMemory;
- mLock.unlock();
+ ssize_t read = 0;
+ Buffer audioBuffer;
- do {
+ while (userSize >= mFrameSize) {
+ audioBuffer.frameCount = userSize / mFrameSize;
- audioBuffer.frameCount = userSize/frameSize();
-
- // By using a wait count corresponding to twice the timeout period in
- // obtainBuffer() we give a chance to recover once for a read timeout
- // (if media_server crashed for instance) before returning a length of
- // 0 bytes read to the client
- status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS));
+ status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever);
if (err < 0) {
- // out of buffers, return #bytes written
- if (err == status_t(NO_MORE_BUFFERS)) {
+ if (read > 0) {
break;
}
- if (err == status_t(TIMED_OUT)) {
- // return partial transfer count
- return read;
- }
return ssize_t(err);
}
size_t bytesRead = audioBuffer.size;
- memcpy(dst, audioBuffer.i8, bytesRead);
-
- dst += bytesRead;
+ memcpy(buffer, audioBuffer.i8, bytesRead);
+ buffer = ((char *) buffer) + bytesRead;
userSize -= bytesRead;
read += bytesRead;
releaseBuffer(&audioBuffer);
- } while (userSize);
+ }
return read;
}
// -------------------------------------------------------------------------
-bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
+nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
{
- Buffer audioBuffer;
- uint32_t frames = mRemainingFrames;
- size_t readSize;
-
mLock.lock();
- // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
- // while we are accessing the cblk
- sp<IAudioRecord> audioRecord = mAudioRecord;
- sp<IMemory> iMem = mCblkMemory;
- audio_track_cblk_t* cblk = mCblk;
- bool active = mActive;
- uint32_t markerPosition = mMarkerPosition;
- uint32_t newPosition = mNewPosition;
- uint32_t user = cblk->user;
- // determine whether a marker callback will be needed, while locked
- bool needMarker = !mMarkerReached && (mMarkerPosition > 0) && (user >= mMarkerPosition);
- if (needMarker) {
- mMarkerReached = true;
+
+ // Can only reference mCblk while locked
+ int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->flags);
+
+ // Check for track invalidation
+ if (flags & CBLK_INVALID) {
+ (void) restoreRecord_l("processAudioBuffer");
+ mLock.unlock();
+ // Run again immediately, but with a new IAudioRecord
+ return 0;
}
- // determine the number of new position callback(s) that will be needed, while locked
+
+ bool active = mActive;
+
+ // Manage overrun callback, must be done under lock to avoid race with releaseBuffer()
+ bool newOverrun = false;
+ if (flags & CBLK_OVERRUN) {
+ if (!mInOverrun) {
+ mInOverrun = true;
+ newOverrun = true;
+ }
+ }
+
+ // Get current position of server
+ size_t position = mProxy->getPosition();
+
+ // Manage marker callback
+ bool markerReached = false;
+ size_t markerPosition = mMarkerPosition;
+ // FIXME fails for wraparound, need 64 bits
+ if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) {
+ mMarkerReached = markerReached = true;
+ }
+
+ // Determine the number of new position callback(s) that will be needed, while locked
+ size_t newPosCount = 0;
+ size_t newPosition = mNewPosition;
uint32_t updatePeriod = mUpdatePeriod;
- uint32_t needNewPos = updatePeriod > 0 && user >= newPosition ?
- ((user - newPosition) / updatePeriod) + 1 : 0;
- mNewPosition = newPosition + updatePeriod * needNewPos;
+ // FIXME fails for wraparound, need 64 bits
+ if (updatePeriod > 0 && position >= newPosition) {
+ newPosCount = ((position - newPosition) / updatePeriod) + 1;
+ mNewPosition += updatePeriod * newPosCount;
+ }
+
+ // Cache other fields that will be needed soon
+ size_t notificationFrames = mNotificationFrames;
+ if (mRefreshRemaining) {
+ mRefreshRemaining = false;
+ mRemainingFrames = notificationFrames;
+ mRetryOnPartialBuffer = false;
+ }
+ size_t misalignment = mProxy->getMisalignment();
+ int32_t sequence = mSequence;
+
+ // These fields don't need to be cached, because they are assigned only by set():
+ // mTransfer, mCbf, mUserData, mSampleRate
+
mLock.unlock();
- // perform marker callback, while unlocked
- if (needMarker) {
+ // perform callbacks while unlocked
+ if (newOverrun) {
+ mCbf(EVENT_OVERRUN, mUserData, NULL);
+ }
+ if (markerReached) {
mCbf(EVENT_MARKER, mUserData, &markerPosition);
}
-
- // perform new position callback(s), while unlocked
- for (; needNewPos > 0; --needNewPos) {
- uint32_t temp = newPosition;
+ while (newPosCount > 0) {
+ size_t temp = newPosition;
mCbf(EVENT_NEW_POS, mUserData, &temp);
newPosition += updatePeriod;
+ newPosCount--;
+ }
+ if (mObservedSequence != sequence) {
+ mObservedSequence = sequence;
+ mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL);
}
- do {
- audioBuffer.frameCount = frames;
- // Calling obtainBuffer() with a wait count of 1
- // limits wait time to WAIT_PERIOD_MS. This prevents from being
- // stuck here not being able to handle timed events (position, markers).
- status_t err = obtainBuffer(&audioBuffer, 1);
- if (err < NO_ERROR) {
- if (err != TIMED_OUT) {
- ALOGE_IF(err != status_t(NO_MORE_BUFFERS),
- "Error obtaining an audio buffer, giving up.");
- return false;
+ // if inactive, then don't run me again until re-started
+ if (!active) {
+ return NS_INACTIVE;
+ }
+
+ // Compute the estimated time until the next timed event (position, markers)
+ uint32_t minFrames = ~0;
+ if (!markerReached && position < markerPosition) {
+ minFrames = markerPosition - position;
+ }
+ if (updatePeriod > 0 && updatePeriod < minFrames) {
+ minFrames = updatePeriod;
+ }
+
+ // If > 0, poll periodically to recover from a stuck server. A good value is 2.
+ static const uint32_t kPoll = 0;
+ if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
+ minFrames = kPoll * notificationFrames;
+ }
+
+ // Convert frame units to time units
+ nsecs_t ns = NS_WHENEVER;
+ if (minFrames != (uint32_t) ~0) {
+ // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
+ static const nsecs_t kFudgeNs = 10000000LL; // 10 ms
+ ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs;
+ }
+
+ // If not supplying data by EVENT_MORE_DATA, then we're done
+ if (mTransfer != TRANSFER_CALLBACK) {
+ return ns;
+ }
+
+ struct timespec timeout;
+ const struct timespec *requested = &ClientProxy::kForever;
+ if (ns != NS_WHENEVER) {
+ timeout.tv_sec = ns / 1000000000LL;
+ timeout.tv_nsec = ns % 1000000000LL;
+ ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
+ requested = &timeout;
+ }
+
+ while (mRemainingFrames > 0) {
+
+ Buffer audioBuffer;
+ audioBuffer.frameCount = mRemainingFrames;
+ size_t nonContig;
+ status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
+ LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
+ "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+ requested = &ClientProxy::kNonBlocking;
+ size_t avail = audioBuffer.frameCount + nonContig;
+ ALOGV("obtainBuffer(%u) returned %u = %u + %u",
+ mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+ if (err != NO_ERROR) {
+ if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
+ break;
}
- break;
+ ALOGE("Error %d obtaining an audio buffer, giving up.", err);
+ return NS_NEVER;
}
- if (err == status_t(STOPPED)) return false;
+
+ if (mRetryOnPartialBuffer) {
+ mRetryOnPartialBuffer = false;
+ if (avail < mRemainingFrames) {
+ int64_t myns = ((mRemainingFrames - avail) *
+ 1100000000LL) / mSampleRate;
+ if (ns < 0 || myns < ns) {
+ ns = myns;
+ }
+ return ns;
+ }
+ }
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
- readSize = audioBuffer.size;
+ size_t readSize = audioBuffer.size;
// Sanity check on returned size
- if (ssize_t(readSize) <= 0) {
- // The callback is done filling buffers
- // Keep this thread going to handle timed events and
- // still try to get more data in intervals of WAIT_PERIOD_MS
- // but don't just loop and block the CPU, so wait
- usleep(WAIT_PERIOD_MS*1000);
- break;
+ if (ssize_t(readSize) < 0 || readSize > reqSize) {
+ ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
+ reqSize, (int) readSize);
+ return NS_NEVER;
}
- if (readSize > reqSize) readSize = reqSize;
- audioBuffer.size = readSize;
- audioBuffer.frameCount = readSize/frameSize();
- frames -= audioBuffer.frameCount;
+ if (readSize == 0) {
+ // The callback is done consuming buffers
+ // Keep this thread going to handle timed events and
+ // still try to provide more data in intervals of WAIT_PERIOD_MS
+ // but don't just loop and block the CPU, so wait
+ return WAIT_PERIOD_MS * 1000000LL;
+ }
+
+ size_t releasedFrames = readSize / mFrameSize;
+ audioBuffer.frameCount = releasedFrames;
+ mRemainingFrames -= releasedFrames;
+ if (misalignment >= releasedFrames) {
+ misalignment -= releasedFrames;
+ } else {
+ misalignment = 0;
+ }
releaseBuffer(&audioBuffer);
- } while (frames);
-
-
- // Manage overrun callback
- if (active && (mProxy->framesAvailable() == 0)) {
- // The value of active is stale, but we are almost sure to be active here because
- // otherwise we would have exited when obtainBuffer returned STOPPED earlier.
- ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
- if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) {
- mCbf(EVENT_OVERRUN, mUserData, NULL);
+ // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
+ // if callback doesn't like to accept the full chunk
+ if (readSize < reqSize) {
+ continue;
}
- }
- if (frames == 0) {
- mRemainingFrames = mNotificationFrames;
- } else {
- mRemainingFrames = frames;
+ // There could be enough non-contiguous frames available to satisfy the remaining request
+ if (mRemainingFrames <= nonContig) {
+ continue;
+ }
+
+#if 0
+ // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
+ // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
+ // that total to a sum == notificationFrames.
+ if (0 < misalignment && misalignment <= mRemainingFrames) {
+ mRemainingFrames = misalignment;
+ return (mRemainingFrames * 1100000000LL) / mSampleRate;
+ }
+#endif
+
}
- return true;
+ mRemainingFrames = notificationFrames;
+ mRetryOnPartialBuffer = true;
+
+ // A lot has transpired since ns was calculated, so run again immediately and re-calculate
+ return 0;
}
-// must be called with mLock and cblk.lock held. Callers must also hold strong references on
-// the IAudioRecord and IMemory in case they are recreated here.
-// If the IAudioRecord is successfully restored, the cblk pointer is updated
-status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& refCblk)
+status_t AudioRecord::restoreRecord_l(const char *from)
{
+ ALOGW("dead IAudioRecord, creating a new one from %s()", from);
+ ++mSequence;
status_t result;
- audio_track_cblk_t* cblk = refCblk;
- audio_track_cblk_t* newCblk = cblk;
- ALOGW("dead IAudioRecord, creating a new one");
-
- // signal old cblk condition so that other threads waiting for available buffers stop
- // waiting now
- cblk->cv.broadcast();
- cblk->lock.unlock();
-
// if the new IAudioRecord is created, openRecord_l() will modify the
// following member variables: mAudioRecord, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioRecord and IMemory
- result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l());
+ size_t position = mProxy->getPosition();
+ mNewPosition = position + mUpdatePeriod;
+ result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l(), position);
if (result == NO_ERROR) {
- newCblk = mCblk;
- // callback thread or sync event hasn't changed
- result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
+ if (mActive) {
+ // callback thread or sync event hasn't changed
+ // FIXME this fails if we have a new AudioFlinger instance
+ result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
+ }
}
if (result != NO_ERROR) {
+ ALOGW("restoreRecord_l() failed status %d", result);
mActive = false;
}
- ALOGV("restoreRecord_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
- result, mActive, newCblk, cblk, newCblk->flags, cblk->flags);
-
- if (result == NO_ERROR) {
- // from now on we switch to the newly created cblk
- refCblk = newCblk;
- }
- newCblk->lock.lock();
-
- ALOGW_IF(result != NO_ERROR, "restoreRecord_l() error %d", result);
-
return result;
}
// =========================================================================
+void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who)
+{
+ sp<AudioRecord> audioRecord = mAudioRecord.promote();
+ if (audioRecord != 0) {
+ AutoMutex lock(audioRecord->mLock);
+ audioRecord->mProxy->binderDied();
+ }
+}
+
+// =========================================================================
+
AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava)
- : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
+ : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false)
{
}
@@ -842,10 +928,26 @@
return true;
}
}
- if (!mReceiver.processAudioBuffer(this)) {
- pause();
+ nsecs_t ns = mReceiver.processAudioBuffer(this);
+ switch (ns) {
+ case 0:
+ return true;
+ case NS_WHENEVER:
+ sleep(1);
+ return true;
+ case NS_INACTIVE:
+ pauseConditional();
+ return true;
+ case NS_NEVER:
+ return false;
+ default:
+ LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+ struct timespec req;
+ req.tv_sec = ns / 1000000000LL;
+ req.tv_nsec = ns % 1000000000LL;
+ nanosleep(&req, NULL /*rem*/);
+ return true;
}
- return true;
}
void AudioRecord::AudioRecordThread::requestExit()
@@ -859,6 +961,17 @@
{
AutoMutex _l(mMyLock);
mPaused = true;
+ mResumeLatch = false;
+}
+
+void AudioRecord::AudioRecordThread::pauseConditional()
+{
+ AutoMutex _l(mMyLock);
+ if (mResumeLatch) {
+ mResumeLatch = false;
+ } else {
+ mPaused = true;
+ }
}
void AudioRecord::AudioRecordThread::resume()
@@ -866,7 +979,10 @@
AutoMutex _l(mMyLock);
if (mPaused) {
mPaused = false;
+ mResumeLatch = false;
mMyCond.signal();
+ } else {
+ mResumeLatch = true;
}
}
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7eeb4f8..faca054 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -19,31 +19,14 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioTrack"
-#include <stdint.h>
-#include <sys/types.h>
-#include <limits.h>
-
-#include <sched.h>
#include <sys/resource.h>
-
+#include <audio_utils/primitives.h>
+#include <binder/IPCThreadState.h>
+#include <media/AudioTrack.h>
+#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
-#include <media/AudioSystem.h>
-#include <media/AudioTrack.h>
-
-#include <utils/Log.h>
-#include <binder/Parcel.h>
-#include <binder/IPCThreadState.h>
-#include <utils/Timers.h>
-#include <utils/Atomic.h>
-
-#include <cutils/bitops.h>
-#include <cutils/compiler.h>
-
-#include <system/audio.h>
-#include <system/audio_policy.h>
-
-#include <audio_utils/primitives.h>
+#define WAIT_PERIOD_MS 10
namespace android {
// ---------------------------------------------------------------------------
@@ -82,7 +65,9 @@
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
- if (minBufCount < 2) minBufCount = 2;
+ if (minBufCount < 2) {
+ minBufCount = 2;
+ }
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
@@ -97,8 +82,7 @@
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT),
- mProxy(NULL)
+ mPreviousSchedulingGroup(SP_DEFAULT)
{
}
@@ -112,16 +96,16 @@
callback_t cbf,
void* user,
int notificationFrames,
- int sessionId)
+ int sessionId,
+ transfer_type transferType)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT),
- mProxy(NULL)
+ mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
- 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
+ 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType);
}
AudioTrack::AudioTrack(
@@ -134,27 +118,20 @@
callback_t cbf,
void* user,
int notificationFrames,
- int sessionId)
+ int sessionId,
+ transfer_type transferType)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT),
- mProxy(NULL)
+ mPreviousSchedulingGroup(SP_DEFAULT)
{
- if (sharedBuffer == 0) {
- ALOGE("sharedBuffer must be non-0");
- mStatus = BAD_VALUE;
- return;
- }
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
- sharedBuffer, false /*threadCanCallJava*/, sessionId);
+ sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType);
}
AudioTrack::~AudioTrack()
{
- ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
-
if (mStatus == NO_ERROR) {
// Make sure that callback function exits in the case where
// it is looping on buffer full condition in obtainBuffer().
@@ -165,11 +142,13 @@
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
- mAudioTrack.clear();
+ if (mAudioTrack != 0) {
+ mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mAudioTrack.clear();
+ }
IPCThreadState::self()->flushCommands();
AudioSystem::releaseAudioSessionId(mSessionId);
}
- delete mProxy;
}
status_t AudioTrack::set(
@@ -184,8 +163,44 @@
int notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
- int sessionId)
+ int sessionId,
+ transfer_type transferType)
{
+ switch (transferType) {
+ case TRANSFER_DEFAULT:
+ if (sharedBuffer != 0) {
+ transferType = TRANSFER_SHARED;
+ } else if (cbf == NULL || threadCanCallJava) {
+ transferType = TRANSFER_SYNC;
+ } else {
+ transferType = TRANSFER_CALLBACK;
+ }
+ break;
+ case TRANSFER_CALLBACK:
+ if (cbf == NULL || sharedBuffer != 0) {
+ ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0");
+ return BAD_VALUE;
+ }
+ break;
+ case TRANSFER_OBTAIN:
+ case TRANSFER_SYNC:
+ if (sharedBuffer != 0) {
+ ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
+ return BAD_VALUE;
+ }
+ break;
+ case TRANSFER_SHARED:
+ if (sharedBuffer == 0) {
+ ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
+ return BAD_VALUE;
+ }
+ break;
+ default:
+ ALOGE("Invalid transfer type %d", transferType);
+ return BAD_VALUE;
+ }
+ mTransfer = transferType;
+
// FIXME "int" here is legacy and will be replaced by size_t later
if (frameCountInt < 0) {
ALOGE("Invalid frame count %d", frameCountInt);
@@ -199,6 +214,7 @@
ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
+
if (mAudioTrack != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
@@ -228,7 +244,7 @@
// validate parameters
if (!audio_is_valid_format(format)) {
- ALOGE("Invalid format");
+ ALOGE("Invalid format %d", format);
return BAD_VALUE;
}
@@ -281,6 +297,7 @@
mFrameCount = frameCount;
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
+ mNotificationFramesAct = 0;
mSessionId = sessionId;
mAuxEffectId = 0;
mFlags = flags;
@@ -298,7 +315,8 @@
frameCount,
flags,
sharedBuffer,
- output);
+ output,
+ 0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
@@ -309,20 +327,21 @@
}
mStatus = NO_ERROR;
-
mStreamType = streamType;
mFormat = format;
-
mSharedBuffer = sharedBuffer;
- mActive = false;
+ mState = STATE_STOPPED;
mUserData = user;
- mLoopCount = 0;
+ mLoopPeriod = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
- mFlushed = false;
AudioSystem::acquireAudioSessionId(mSessionId);
+ mSequence = 1;
+ mObservedSequence = mSequence;
+ mInUnderrun = false;
+
return NO_ERROR;
}
@@ -330,87 +349,45 @@
void AudioTrack::start()
{
- sp<AudioTrackThread> t = mAudioTrackThread;
-
- ALOGV("start %p", this);
-
AutoMutex lock(mLock);
- // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
- // while we are accessing the cblk
- sp<IAudioTrack> audioTrack = mAudioTrack;
- sp<IMemory> iMem = mCblkMemory;
- audio_track_cblk_t* cblk = mCblk;
-
- if (!mActive) {
- mFlushed = false;
- mActive = true;
- mNewPosition = cblk->server + mUpdatePeriod;
- cblk->lock.lock();
- cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
- cblk->waitTimeMs = 0;
- android_atomic_and(~CBLK_DISABLED, &cblk->flags);
- if (t != 0) {
- t->resume();
- } else {
- mPreviousPriority = getpriority(PRIO_PROCESS, 0);
- get_sched_policy(0, &mPreviousSchedulingGroup);
- androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
- }
-
- ALOGV("start %p before lock cblk %p", this, cblk);
- status_t status = NO_ERROR;
- if (!(cblk->flags & CBLK_INVALID)) {
- cblk->lock.unlock();
- ALOGV("mAudioTrack->start()");
- status = mAudioTrack->start();
- cblk->lock.lock();
- if (status == DEAD_OBJECT) {
- android_atomic_or(CBLK_INVALID, &cblk->flags);
- }
- }
- if (cblk->flags & CBLK_INVALID) {
- audio_track_cblk_t* temp = cblk;
- status = restoreTrack_l(temp, true /*fromStart*/);
- cblk = temp;
- }
- cblk->lock.unlock();
- if (status != NO_ERROR) {
- ALOGV("start() failed");
- mActive = false;
- if (t != 0) {
- t->pause();
- } else {
- setpriority(PRIO_PROCESS, 0, mPreviousPriority);
- set_sched_policy(0, mPreviousSchedulingGroup);
- }
- }
+ if (mState == STATE_ACTIVE) {
+ return;
}
-}
+ mInUnderrun = true;
-void AudioTrack::stop()
-{
+ State previousState = mState;
+ mState = STATE_ACTIVE;
+ if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
+ // reset current position as seen by client to 0
+ mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
+ }
+ mNewPosition = mProxy->getPosition() + mUpdatePeriod;
+ int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->flags);
+
sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->resume();
+ } else {
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ get_sched_policy(0, &mPreviousSchedulingGroup);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
+ }
- ALOGV("stop %p", this);
-
- AutoMutex lock(mLock);
- if (mActive) {
- mActive = false;
- mCblk->cv.signal();
- mAudioTrack->stop();
- // Cancel loops (If we are in the middle of a loop, playback
- // would not stop until loopCount reaches 0).
- setLoop_l(0, 0, 0);
- // the playback head position will reset to 0, so if a marker is set, we need
- // to activate it again
- mMarkerReached = false;
- // Force flush if a shared buffer is used otherwise audioflinger
- // will not stop before end of buffer is reached.
- // It may be needed to make sure that we stop playback, likely in case looping is on.
- if (mSharedBuffer != 0) {
- flush_l();
+ status_t status = NO_ERROR;
+ if (!(flags & CBLK_INVALID)) {
+ status = mAudioTrack->start();
+ if (status == DEAD_OBJECT) {
+ flags |= CBLK_INVALID;
}
+ }
+ if (flags & CBLK_INVALID) {
+ status = restoreTrack_l("start");
+ }
+
+ if (status != NO_ERROR) {
+ ALOGE("start() status %d", status);
+ mState = previousState;
if (t != 0) {
t->pause();
} else {
@@ -419,57 +396,85 @@
}
}
+ // FIXME discarding status
+}
+
+void AudioTrack::stop()
+{
+ AutoMutex lock(mLock);
+ // FIXME pause then stop should not be a nop
+ if (mState != STATE_ACTIVE) {
+ return;
+ }
+
+ mState = STATE_STOPPED;
+ mProxy->interrupt();
+ mAudioTrack->stop();
+ // the playback head position will reset to 0, so if a marker is set, we need
+ // to activate it again
+ mMarkerReached = false;
+#if 0
+ // Force flush if a shared buffer is used otherwise audioflinger
+ // will not stop before end of buffer is reached.
+ // It may be needed to make sure that we stop playback, likely in case looping is on.
+ if (mSharedBuffer != 0) {
+ flush_l();
+ }
+#endif
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->pause();
+ } else {
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ set_sched_policy(0, mPreviousSchedulingGroup);
+ }
}
bool AudioTrack::stopped() const
{
AutoMutex lock(mLock);
- return stopped_l();
+ return mState != STATE_ACTIVE;
}
void AudioTrack::flush()
{
- AutoMutex lock(mLock);
- if (!mActive && mSharedBuffer == 0) {
- flush_l();
+ if (mSharedBuffer != 0) {
+ return;
}
+ AutoMutex lock(mLock);
+ if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) {
+ return;
+ }
+ flush_l();
}
void AudioTrack::flush_l()
{
- ALOGV("flush");
- ALOG_ASSERT(!mActive);
+ ALOG_ASSERT(mState != STATE_ACTIVE);
// clear playback marker and periodic update counter
mMarkerPosition = 0;
mMarkerReached = false;
mUpdatePeriod = 0;
- mFlushed = true;
+ mState = STATE_FLUSHED;
+ mProxy->flush();
mAudioTrack->flush();
- // Release AudioTrack callback thread in case it was waiting for new buffers
- // in AudioTrack::obtainBuffer()
- mCblk->cv.signal();
}
void AudioTrack::pause()
{
- ALOGV("pause");
AutoMutex lock(mLock);
- if (mActive) {
- mActive = false;
- mCblk->cv.signal();
- mAudioTrack->pause();
+ if (mState != STATE_ACTIVE) {
+ return;
}
+ mState = STATE_PAUSED;
+ mProxy->interrupt();
+ mAudioTrack->pause();
}
status_t AudioTrack::setVolume(float left, float right)
{
- if (mStatus != NO_ERROR) {
- return mStatus;
- }
- ALOG_ASSERT(mProxy != NULL);
-
if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
return BAD_VALUE;
}
@@ -490,18 +495,11 @@
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
- ALOGV("setAuxEffectSendLevel(%f)", level);
-
- if (mStatus != NO_ERROR) {
- return mStatus;
- }
- ALOG_ASSERT(mProxy != NULL);
-
if (level < 0.0f || level > 1.0f) {
return BAD_VALUE;
}
- AutoMutex lock(mLock);
+ AutoMutex lock(mLock);
mSendLevel = level;
mProxy->setSendLevel(level);
@@ -511,18 +509,17 @@
void AudioTrack::getAuxEffectSendLevel(float* level) const
{
if (level != NULL) {
- *level = mSendLevel;
+ *level = mSendLevel;
}
}
status_t AudioTrack::setSampleRate(uint32_t rate)
{
- uint32_t afSamplingRate;
-
if (mIsTimed) {
return INVALID_OPERATION;
}
+ uint32_t afSamplingRate;
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return NO_INIT;
}
@@ -550,58 +547,44 @@
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
- AutoMutex lock(mLock);
- return setLoop_l(loopStart, loopEnd, loopCount);
-}
-
-// must be called with mLock held
-status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
-{
if (mSharedBuffer == 0 || mIsTimed) {
return INVALID_OPERATION;
}
- audio_track_cblk_t* cblk = mCblk;
-
- Mutex::Autolock _l(cblk->lock);
-
if (loopCount == 0) {
- cblk->loopStart = UINT_MAX;
- cblk->loopEnd = UINT_MAX;
- cblk->loopCount = 0;
- mLoopCount = 0;
- return NO_ERROR;
- }
-
- if (loopStart >= loopEnd ||
- loopEnd - loopStart > mFrameCount ||
- cblk->server > loopStart) {
- ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, "
- "user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
+ ;
+ } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
+ loopEnd - loopStart >= MIN_LOOP) {
+ ;
+ } else {
return BAD_VALUE;
}
- if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) {
- ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, "
- "framecount %d",
- loopStart, loopEnd, mFrameCount);
- return BAD_VALUE;
+ AutoMutex lock(mLock);
+ // See setPosition() regarding setting parameters such as loop points or position while active
+ if (mState == STATE_ACTIVE) {
+ return INVALID_OPERATION;
}
-
- cblk->loopStart = loopStart;
- cblk->loopEnd = loopEnd;
- cblk->loopCount = loopCount;
- mLoopCount = loopCount;
-
+ setLoop_l(loopStart, loopEnd, loopCount);
return NO_ERROR;
}
+void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
+{
+ // FIXME If setting a loop also sets position to start of loop, then
+ // this is correct. Otherwise it should be removed.
+ mNewPosition = mProxy->getPosition() + mUpdatePeriod;
+ mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0;
+ mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
+}
+
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
if (mCbf == NULL) {
return INVALID_OPERATION;
}
+ AutoMutex lock(mLock);
mMarkerPosition = marker;
mMarkerReached = false;
@@ -614,6 +597,7 @@
return BAD_VALUE;
}
+ AutoMutex lock(mLock);
*marker = mMarkerPosition;
return NO_ERROR;
@@ -625,9 +609,8 @@
return INVALID_OPERATION;
}
- uint32_t curPosition;
- getPosition(&curPosition);
- mNewPosition = curPosition + updatePeriod;
+ AutoMutex lock(mLock);
+ mNewPosition = mProxy->getPosition() + updatePeriod;
mUpdatePeriod = updatePeriod;
return NO_ERROR;
@@ -639,6 +622,7 @@
return BAD_VALUE;
}
+ AutoMutex lock(mLock);
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
@@ -649,58 +633,73 @@
if (mSharedBuffer == 0 || mIsTimed) {
return INVALID_OPERATION;
}
-
- AutoMutex lock(mLock);
-
- if (!stopped_l()) {
- return INVALID_OPERATION;
- }
-
- audio_track_cblk_t* cblk = mCblk;
- Mutex::Autolock _l(cblk->lock);
-
- if (position > cblk->user) {
+ if (position > mFrameCount) {
return BAD_VALUE;
}
- cblk->server = position;
- android_atomic_or(CBLK_FORCEREADY, &cblk->flags);
+ AutoMutex lock(mLock);
+ // Currently we require that the player is inactive before setting parameters such as position
+ // or loop points. Otherwise, there could be a race condition: the application could read the
+ // current position, compute a new position or loop parameters, and then set that position or
+ // loop parameters but it would do the "wrong" thing since the position has continued to advance
+ // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
+ // to specify how it wants to handle such scenarios.
+ if (mState == STATE_ACTIVE) {
+ return INVALID_OPERATION;
+ }
+ mNewPosition = mProxy->getPosition() + mUpdatePeriod;
+ mLoopPeriod = 0;
+ // FIXME Check whether loops and setting position are incompatible in old code.
+ // If we use setLoop for both purposes we lose the capability to set the position while looping.
+ mStaticProxy->setLoop(position, mFrameCount, 0);
return NO_ERROR;
}
-status_t AudioTrack::getPosition(uint32_t *position)
+status_t AudioTrack::getPosition(uint32_t *position) const
{
if (position == NULL) {
return BAD_VALUE;
}
- AutoMutex lock(mLock);
- *position = mFlushed ? 0 : mCblk->server;
+ AutoMutex lock(mLock);
+ // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
+ *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 :
+ mProxy->getPosition();
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getBufferPosition(size_t *position)
+{
+ if (mSharedBuffer == 0 || mIsTimed) {
+ return INVALID_OPERATION;
+ }
+ if (position == NULL) {
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mLock);
+ *position = mStaticProxy->getBufferPosition();
return NO_ERROR;
}
status_t AudioTrack::reload()
{
- if (mStatus != NO_ERROR) {
- return mStatus;
- }
- ALOG_ASSERT(mProxy != NULL);
-
if (mSharedBuffer == 0 || mIsTimed) {
return INVALID_OPERATION;
}
AutoMutex lock(mLock);
-
- if (!stopped_l()) {
+ // See setPosition() regarding setting parameters such as loop points or position while active
+ if (mState == STATE_ACTIVE) {
return INVALID_OPERATION;
}
-
- flush_l();
-
- (void) mProxy->stepUser(mFrameCount);
-
+ mNewPosition = mUpdatePeriod;
+ mLoopPeriod = 0;
+ // FIXME The new code cannot reload while keeping a loop specified.
+ // Need to check how the old code handled this, and whether it's a significant change.
+ mStaticProxy->setLoop(0, mFrameCount, 0);
return NO_ERROR;
}
@@ -719,7 +718,7 @@
status_t AudioTrack::attachAuxEffect(int effectId)
{
- ALOGV("attachAuxEffect(%d)", effectId);
+ AutoMutex lock(mLock);
status_t status = mAudioTrack->attachAuxEffect(effectId);
if (status == NO_ERROR) {
mAuxEffectId = effectId;
@@ -737,7 +736,8 @@
size_t frameCount,
audio_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output)
+ audio_io_handle_t output,
+ size_t epoch)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -747,7 +747,8 @@
}
uint32_t afLatency;
- if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
+ if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) {
+ ALOGE("getLatency(%d) failed status %d", output, status);
return NO_INIT;
}
@@ -775,7 +776,10 @@
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
size_t afFrameCount;
- if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
+ status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+ if (status != NO_ERROR) {
+ ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType,
+ status);
return NO_INIT;
}
frameCount = afFrameCount;
@@ -806,17 +810,26 @@
// FIXME move these calculations and associated checks to server
uint32_t afSampleRate;
- if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
+ status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
+ if (status != NO_ERROR) {
+ ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType,
+ status);
return NO_INIT;
}
size_t afFrameCount;
- if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
+ status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+ if (status != NO_ERROR) {
+ ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
return NO_INIT;
}
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
- if (minBufCount < 2) minBufCount = 2;
+ ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
+ afFrameCount, minBufCount, afSampleRate, afLatency);
+ if (minBufCount <= 2) {
+ minBufCount = sampleRate == afSampleRate ? 2 : 3;
+ }
size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
@@ -826,12 +839,9 @@
if (frameCount == 0) {
frameCount = minFrameCount;
}
- if (mNotificationFramesAct == 0) {
- mNotificationFramesAct = frameCount/2;
- }
// Make sure that application is notified with sufficient margin
// before underrun
- if (mNotificationFramesAct > frameCount/2) {
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
mNotificationFramesAct = frameCount/2;
}
if (frameCount < minFrameCount) {
@@ -881,6 +891,10 @@
ALOGE("Could not get control block");
return NO_INIT;
}
+ if (mAudioTrack != 0) {
+ mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mDeathNotifier.clear();
+ }
mAudioTrack = track;
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
@@ -898,26 +912,38 @@
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
mAwaitBoost = true;
+ if (sharedBuffer == 0) {
+ // double-buffering is not required for fast tracks, due to tighter scheduling
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) {
+ mNotificationFramesAct = frameCount;
+ }
+ }
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
// once denied, do not request again if IAudioTrack is re-created
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
- }
- if (sharedBuffer == 0) {
- mNotificationFramesAct = frameCount/2;
+ if (sharedBuffer == 0) {
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
+ mNotificationFramesAct = frameCount/2;
+ }
+ }
}
}
+ mRefreshRemaining = true;
+
+ // Starting address of buffers in shared memory. If there is a shared buffer, buffers
+ // is the value of pointer() for the shared buffer, otherwise buffers points
+ // immediately after the control block. This address is for the mapping within client
+ // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
+ void* buffers;
if (sharedBuffer == 0) {
- mBuffers = (char*)cblk + sizeof(audio_track_cblk_t);
+ buffers = (char*)cblk + sizeof(audio_track_cblk_t);
} else {
- mBuffers = sharedBuffer->pointer();
+ buffers = sharedBuffer->pointer();
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
- cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
- cblk->waitTimeMs = 0;
- mRemainingFrames = mNotificationFramesAct;
// FIXME don't believe this lie
mLatency = afLatency + (1000*frameCount) / sampleRate;
mFrameCount = frameCount;
@@ -928,147 +954,143 @@
}
// update proxy
- delete mProxy;
- mProxy = new AudioTrackClientProxy(cblk, mBuffers, frameCount, mFrameSizeAF);
+ if (sharedBuffer == 0) {
+ mStaticProxy.clear();
+ mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ } else {
+ mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ mProxy = mStaticProxy;
+ }
mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
uint16_t(mVolume[LEFT] * 0x1000));
mProxy->setSendLevel(mSendLevel);
mProxy->setSampleRate(mSampleRate);
- if (sharedBuffer != 0) {
- // Force buffer full condition as data is already present in shared memory
- mProxy->stepUser(frameCount);
- }
+ mProxy->setEpoch(epoch);
+ mProxy->setMinimum(mNotificationFramesAct);
+
+ mDeathNotifier = new DeathNotifier(this);
+ mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
- ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
-
- AutoMutex lock(mLock);
- bool active;
- status_t result = NO_ERROR;
- audio_track_cblk_t* cblk = mCblk;
- uint32_t framesReq = audioBuffer->frameCount;
- uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
-
- audioBuffer->frameCount = 0;
- audioBuffer->size = 0;
-
- size_t framesAvail = mProxy->framesAvailable();
-
- cblk->lock.lock();
- if (cblk->flags & CBLK_INVALID) {
- goto create_new_track;
+ if (audioBuffer == NULL) {
+ return BAD_VALUE;
}
- cblk->lock.unlock();
+ if (mTransfer != TRANSFER_OBTAIN) {
+ audioBuffer->frameCount = 0;
+ audioBuffer->size = 0;
+ audioBuffer->raw = NULL;
+ return INVALID_OPERATION;
+ }
- if (framesAvail == 0) {
- cblk->lock.lock();
- goto start_loop_here;
- while (framesAvail == 0) {
- active = mActive;
- if (CC_UNLIKELY(!active)) {
- ALOGV("Not active and NO_MORE_BUFFERS");
- cblk->lock.unlock();
- return NO_MORE_BUFFERS;
- }
- if (CC_UNLIKELY(!waitCount)) {
- cblk->lock.unlock();
- return WOULD_BLOCK;
- }
- if (!(cblk->flags & CBLK_INVALID)) {
- mLock.unlock();
- // this condition is in shared memory, so if IAudioTrack and control block
- // are replaced due to mediaserver death or IAudioTrack invalidation then
- // cv won't be signalled, but fortunately the timeout will limit the wait
- result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
- cblk->lock.unlock();
- mLock.lock();
- if (!mActive) {
- return status_t(STOPPED);
- }
- // IAudioTrack may have been re-created while mLock was unlocked
- cblk = mCblk;
- cblk->lock.lock();
- }
+ const struct timespec *requested;
+ if (waitCount == -1) {
+ requested = &ClientProxy::kForever;
+ } else if (waitCount == 0) {
+ requested = &ClientProxy::kNonBlocking;
+ } else if (waitCount > 0) {
+ long long ms = WAIT_PERIOD_MS * (long long) waitCount;
+ struct timespec timeout;
+ timeout.tv_sec = ms / 1000;
+ timeout.tv_nsec = (int) (ms % 1000) * 1000000;
+ requested = &timeout;
+ } else {
+ ALOGE("%s invalid waitCount %d", __func__, waitCount);
+ requested = NULL;
+ }
+ return obtainBuffer(audioBuffer, requested);
+}
- if (cblk->flags & CBLK_INVALID) {
- goto create_new_track;
- }
- if (CC_UNLIKELY(result != NO_ERROR)) {
- cblk->waitTimeMs += waitTimeMs;
- if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
- // timing out when a loop has been set and we have already written upto loop end
- // is a normal condition: no need to wake AudioFlinger up.
- if (cblk->user < cblk->loopEnd) {
- ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, "
- "server=%08x", this, cblk->mName, cblk->user, cblk->server);
- //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
- cblk->lock.unlock();
- result = mAudioTrack->start();
- cblk->lock.lock();
- if (result == DEAD_OBJECT) {
- android_atomic_or(CBLK_INVALID, &cblk->flags);
-create_new_track:
- audio_track_cblk_t* temp = cblk;
- result = restoreTrack_l(temp, false /*fromStart*/);
- cblk = temp;
- }
- if (result != NO_ERROR) {
- ALOGW("obtainBuffer create Track error %d", result);
- cblk->lock.unlock();
- return result;
- }
+status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
+ struct timespec *elapsed, size_t *nonContig)
+{
+ // previous and new IAudioTrack sequence numbers are used to detect track re-creation
+ uint32_t oldSequence = 0;
+ uint32_t newSequence;
+
+ Proxy::Buffer buffer;
+ status_t status = NO_ERROR;
+
+ static const int32_t kMaxTries = 5;
+ int32_t tryCounter = kMaxTries;
+
+ do {
+ // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
+ // keep them from going away if another thread re-creates the track during obtainBuffer()
+ sp<AudioTrackClientProxy> proxy;
+ sp<IMemory> iMem;
+
+ { // start of lock scope
+ AutoMutex lock(mLock);
+
+ newSequence = mSequence;
+ // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
+ if (status == DEAD_OBJECT) {
+ // re-create track, unless someone else has already done so
+ if (newSequence == oldSequence) {
+ status = restoreTrack_l("obtainBuffer");
+ if (status != NO_ERROR) {
+ break;
}
- cblk->waitTimeMs = 0;
- }
-
- if (--waitCount == 0) {
- cblk->lock.unlock();
- return TIMED_OUT;
}
}
- // read the server count again
- start_loop_here:
- framesAvail = mProxy->framesAvailable_l();
- }
- cblk->lock.unlock();
+ oldSequence = newSequence;
+
+ // Keep the extra references
+ proxy = mProxy;
+ iMem = mCblkMemory;
+
+ // Non-blocking if track is stopped or paused
+ if (mState != STATE_ACTIVE) {
+ requested = &ClientProxy::kNonBlocking;
+ }
+
+ } // end of lock scope
+
+ buffer.mFrameCount = audioBuffer->frameCount;
+ // FIXME starts the requested timeout and elapsed over from scratch
+ status = proxy->obtainBuffer(&buffer, requested, elapsed);
+
+ } while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
+
+ audioBuffer->frameCount = buffer.mFrameCount;
+ audioBuffer->size = buffer.mFrameCount * mFrameSizeAF;
+ audioBuffer->raw = buffer.mRaw;
+ if (nonContig != NULL) {
+ *nonContig = buffer.mNonContig;
}
-
- cblk->waitTimeMs = 0;
-
- if (framesReq > framesAvail) {
- framesReq = framesAvail;
- }
-
- uint32_t u = cblk->user;
- uint32_t bufferEnd = cblk->userBase + mFrameCount;
-
- if (framesReq > bufferEnd - u) {
- framesReq = bufferEnd - u;
- }
-
- audioBuffer->frameCount = framesReq;
- audioBuffer->size = framesReq * mFrameSizeAF;
- audioBuffer->raw = mProxy->buffer(u);
- active = mActive;
- return active ? status_t(NO_ERROR) : status_t(STOPPED);
+ return status;
}
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
- ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL);
+ if (mTransfer == TRANSFER_SHARED) {
+ return;
+ }
+
+ size_t stepCount = audioBuffer->size / mFrameSizeAF;
+ if (stepCount == 0) {
+ return;
+ }
+
+ Proxy::Buffer buffer;
+ buffer.mFrameCount = stepCount;
+ buffer.mRaw = audioBuffer->raw;
AutoMutex lock(mLock);
- audio_track_cblk_t* cblk = mCblk;
- (void) mProxy->stepUser(audioBuffer->frameCount);
- if (audioBuffer->frameCount > 0) {
- // restart track if it was disabled by audioflinger due to previous underrun
- if (mActive && (cblk->flags & CBLK_DISABLED)) {
- android_atomic_and(~CBLK_DISABLED, &cblk->flags);
- ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName);
+ mInUnderrun = false;
+ mProxy->releaseBuffer(&buffer);
+
+ // restart track if it was disabled by audioflinger due to previous underrun
+ if (mState == STATE_ACTIVE) {
+ audio_track_cblk_t* cblk = mCblk;
+ if (android_atomic_and(~CBLK_DISABLED, &cblk->flags) & CBLK_DISABLED) {
+ ALOGW("releaseBuffer() track %p name=%#x disabled due to previous underrun, restarting",
+ this, cblk->mName);
+ // FIXME ignoring status
mAudioTrack->start();
}
}
@@ -1078,68 +1100,46 @@
ssize_t AudioTrack::write(const void* buffer, size_t userSize)
{
-
- if (mSharedBuffer != 0 || mIsTimed) {
+ if (mTransfer != TRANSFER_SYNC || mIsTimed) {
return INVALID_OPERATION;
}
- if (ssize_t(userSize) < 0) {
+ if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
- ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
- buffer, userSize, userSize);
+ ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
return BAD_VALUE;
}
- ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
-
- if (userSize == 0) {
- return 0;
- }
-
- // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
- // while we are accessing the cblk
- mLock.lock();
- sp<IAudioTrack> audioTrack = mAudioTrack;
- sp<IMemory> iMem = mCblkMemory;
- mLock.unlock();
-
- // since mLock is unlocked the IAudioTrack and shared memory may be re-created,
- // so all cblk references might still refer to old shared memory, but that should be benign
-
- ssize_t written = 0;
- const int8_t *src = (const int8_t *)buffer;
+ size_t written = 0;
Buffer audioBuffer;
- size_t frameSz = frameSize();
- do {
- audioBuffer.frameCount = userSize/frameSz;
+ while (userSize >= mFrameSize) {
+ audioBuffer.frameCount = userSize / mFrameSize;
- status_t err = obtainBuffer(&audioBuffer, -1);
+ status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever);
if (err < 0) {
- // out of buffers, return #bytes written
- if (err == status_t(NO_MORE_BUFFERS)) {
+ if (written > 0) {
break;
}
return ssize_t(err);
}
size_t toWrite;
-
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
- toWrite = audioBuffer.size>>1;
- memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
+ toWrite = audioBuffer.size >> 1;
+ memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite);
} else {
toWrite = audioBuffer.size;
- memcpy(audioBuffer.i8, src, toWrite);
+ memcpy(audioBuffer.i8, buffer, toWrite);
}
- src += toWrite;
+ buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
- } while (userSize >= frameSz);
+ }
return written;
}
@@ -1155,10 +1155,12 @@
AutoMutex lock(mLock);
status_t result = UNKNOWN_ERROR;
+#if 1
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
sp<IAudioTrack> audioTrack = mAudioTrack;
sp<IMemory> iMem = mCblkMemory;
+#endif
// If the track is not invalid already, try to allocate a buffer. alloc
// fails indicating that the server is dead, flag the track as invalid so
@@ -1174,13 +1176,9 @@
// If the track is invalid at this point, attempt to restore it. and try the
// allocation one more time.
if (cblk->flags & CBLK_INVALID) {
- cblk->lock.lock();
- audio_track_cblk_t* temp = cblk;
- result = restoreTrack_l(temp, false /*fromStart*/);
- cblk = temp;
- cblk->lock.unlock();
+ result = restoreTrack_l("allocateTimedBuffer");
- if (result == OK) {
+ if (result == NO_ERROR) {
result = mAudioTrack->allocateTimedBuffer(size, buffer);
}
}
@@ -1197,9 +1195,10 @@
audio_track_cblk_t* cblk = mCblk;
// restart track if it was disabled by audioflinger due to previous underrun
if (buffer->size() != 0 && status == NO_ERROR &&
- mActive && (cblk->flags & CBLK_DISABLED)) {
+ (mState == STATE_ACTIVE) && (cblk->flags & CBLK_DISABLED)) {
android_atomic_and(~CBLK_DISABLED, &cblk->flags);
ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
+ // FIXME ignoring status
mAudioTrack->start();
}
}
@@ -1214,12 +1213,8 @@
// -------------------------------------------------------------------------
-bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
+nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
- Buffer audioBuffer;
- uint32_t frames;
- size_t writtenSize;
-
mLock.lock();
if (mAwaitBoost) {
mAwaitBoost = false;
@@ -1240,86 +1235,181 @@
}
return true;
}
- // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
- // while we are accessing the cblk
- sp<IAudioTrack> audioTrack = mAudioTrack;
- sp<IMemory> iMem = mCblkMemory;
- audio_track_cblk_t* cblk = mCblk;
- bool active = mActive;
- mLock.unlock();
- // since mLock is unlocked the IAudioTrack and shared memory may be re-created,
- // so all cblk references might still refer to old shared memory, but that should be benign
+ // Can only reference mCblk while locked
+ int32_t flags = android_atomic_and(
+ ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->flags);
- // Manage underrun callback
- if (active && (mProxy->framesAvailable() == mFrameCount)) {
- ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
- if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) {
- mCbf(EVENT_UNDERRUN, mUserData, 0);
- if (cblk->server == mFrameCount) {
- mCbf(EVENT_BUFFER_END, mUserData, 0);
- }
- if (mSharedBuffer != 0) {
- return false;
- }
+ // Check for track invalidation
+ if (flags & CBLK_INVALID) {
+ (void) restoreTrack_l("processAudioBuffer");
+ mLock.unlock();
+ // Run again immediately, but with a new IAudioTrack
+ return 0;
+ }
+
+ bool active = mState == STATE_ACTIVE;
+
+ // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
+ bool newUnderrun = false;
+ if (flags & CBLK_UNDERRUN) {
+#if 0
+ // Currently in shared buffer mode, when the server reaches the end of buffer,
+ // the track stays active in continuous underrun state. It's up to the application
+ // to pause or stop the track, or set the position to a new offset within buffer.
+ // This was some experimental code to auto-pause on underrun. Keeping it here
+ // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
+ if (mTransfer == TRANSFER_SHARED) {
+ mState = STATE_PAUSED;
+ active = false;
+ }
+#endif
+ if (!mInUnderrun) {
+ mInUnderrun = true;
+ newUnderrun = true;
}
}
- // Manage loop end callback
- while (mLoopCount > cblk->loopCount) {
- int loopCount = -1;
- mLoopCount--;
- if (mLoopCount >= 0) loopCount = mLoopCount;
-
- mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
- }
+ // Get current position of server
+ size_t position = mProxy->getPosition();
// Manage marker callback
- if (!mMarkerReached && (mMarkerPosition > 0)) {
- if (cblk->server >= mMarkerPosition) {
- mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
- mMarkerReached = true;
- }
+ bool markerReached = false;
+ size_t markerPosition = mMarkerPosition;
+ // FIXME fails for wraparound, need 64 bits
+ if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) {
+ mMarkerReached = markerReached = true;
}
- // Manage new position callback
- if (mUpdatePeriod > 0) {
- while (cblk->server >= mNewPosition) {
- mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
- mNewPosition += mUpdatePeriod;
- }
+ // Determine number of new position callback(s) that will be needed, while locked
+ size_t newPosCount = 0;
+ size_t newPosition = mNewPosition;
+ size_t updatePeriod = mUpdatePeriod;
+ // FIXME fails for wraparound, need 64 bits
+ if (updatePeriod > 0 && position >= newPosition) {
+ newPosCount = ((position - newPosition) / updatePeriod) + 1;
+ mNewPosition += updatePeriod * newPosCount;
}
- // If Shared buffer is used, no data is requested from client.
- if (mSharedBuffer != 0) {
- frames = 0;
- } else {
- frames = mRemainingFrames;
+ // Cache other fields that will be needed soon
+ uint32_t loopPeriod = mLoopPeriod;
+ uint32_t sampleRate = mSampleRate;
+ size_t notificationFrames = mNotificationFramesAct;
+ if (mRefreshRemaining) {
+ mRefreshRemaining = false;
+ mRemainingFrames = notificationFrames;
+ mRetryOnPartialBuffer = false;
+ }
+ size_t misalignment = mProxy->getMisalignment();
+ int32_t sequence = mSequence;
+
+ // These fields don't need to be cached, because they are assigned only by set():
+ // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags
+ // mFlags is also assigned by createTrack_l(), but not the bit we care about.
+
+ mLock.unlock();
+
+ // perform callbacks while unlocked
+ if (newUnderrun) {
+ mCbf(EVENT_UNDERRUN, mUserData, NULL);
+ }
+ // FIXME we will miss loops if loop cycle was signaled several times since last call
+ // to processAudioBuffer()
+ if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) {
+ mCbf(EVENT_LOOP_END, mUserData, NULL);
+ }
+ if (flags & CBLK_BUFFER_END) {
+ mCbf(EVENT_BUFFER_END, mUserData, NULL);
+ }
+ if (markerReached) {
+ mCbf(EVENT_MARKER, mUserData, &markerPosition);
+ }
+ while (newPosCount > 0) {
+ size_t temp = newPosition;
+ mCbf(EVENT_NEW_POS, mUserData, &temp);
+ newPosition += updatePeriod;
+ newPosCount--;
+ }
+ if (mObservedSequence != sequence) {
+ mObservedSequence = sequence;
+ mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
}
- // See description of waitCount parameter at declaration of obtainBuffer().
- // The logic below prevents us from being stuck below at obtainBuffer()
- // not being able to handle timed events (position, markers, loops).
- int32_t waitCount = -1;
- if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
- waitCount = 1;
+ // if inactive, then don't run me again until re-started
+ if (!active) {
+ return NS_INACTIVE;
}
- do {
+ // Compute the estimated time until the next timed event (position, markers, loops)
+ // FIXME only for non-compressed audio
+ uint32_t minFrames = ~0;
+ if (!markerReached && position < markerPosition) {
+ minFrames = markerPosition - position;
+ }
+ if (loopPeriod > 0 && loopPeriod < minFrames) {
+ minFrames = loopPeriod;
+ }
+ if (updatePeriod > 0 && updatePeriod < minFrames) {
+ minFrames = updatePeriod;
+ }
- audioBuffer.frameCount = frames;
+ // If > 0, poll periodically to recover from a stuck server. A good value is 2.
+ static const uint32_t kPoll = 0;
+ if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
+ minFrames = kPoll * notificationFrames;
+ }
- status_t err = obtainBuffer(&audioBuffer, waitCount);
- if (err < NO_ERROR) {
- if (err != TIMED_OUT) {
- ALOGE_IF(err != status_t(NO_MORE_BUFFERS),
- "Error obtaining an audio buffer, giving up.");
- return false;
+ // Convert frame units to time units
+ nsecs_t ns = NS_WHENEVER;
+ if (minFrames != (uint32_t) ~0) {
+ // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
+ static const nsecs_t kFudgeNs = 10000000LL; // 10 ms
+ ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs;
+ }
+
+ // If not supplying data by EVENT_MORE_DATA, then we're done
+ if (mTransfer != TRANSFER_CALLBACK) {
+ return ns;
+ }
+
+ struct timespec timeout;
+ const struct timespec *requested = &ClientProxy::kForever;
+ if (ns != NS_WHENEVER) {
+ timeout.tv_sec = ns / 1000000000LL;
+ timeout.tv_nsec = ns % 1000000000LL;
+ ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
+ requested = &timeout;
+ }
+
+ while (mRemainingFrames > 0) {
+
+ Buffer audioBuffer;
+ audioBuffer.frameCount = mRemainingFrames;
+ size_t nonContig;
+ status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
+ LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
+ "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+ requested = &ClientProxy::kNonBlocking;
+ size_t avail = audioBuffer.frameCount + nonContig;
+ ALOGV("obtainBuffer(%u) returned %u = %u + %u",
+ mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+ if (err != NO_ERROR) {
+ if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
+ return 0;
}
- break;
+ ALOGE("Error %d obtaining an audio buffer, giving up.", err);
+ return NS_NEVER;
}
- if (err == status_t(STOPPED)) {
- return false;
+
+ if (mRetryOnPartialBuffer) {
+ mRetryOnPartialBuffer = false;
+ if (avail < mRemainingFrames) {
+ int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate;
+ if (ns < 0 || myns < ns) {
+ ns = myns;
+ }
+ return ns;
+ }
}
// Divide buffer size by 2 to take into account the expansion
@@ -1331,66 +1421,76 @@
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
- writtenSize = audioBuffer.size;
+ size_t writtenSize = audioBuffer.size;
+ size_t writtenFrames = writtenSize / mFrameSize;
// Sanity check on returned size
- if (ssize_t(writtenSize) <= 0) {
+ if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
+ ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
+ reqSize, (int) writtenSize);
+ return NS_NEVER;
+ }
+
+ if (writtenSize == 0) {
// The callback is done filling buffers
// Keep this thread going to handle timed events and
// still try to get more data in intervals of WAIT_PERIOD_MS
// but don't just loop and block the CPU, so wait
- usleep(WAIT_PERIOD_MS*1000);
- break;
- }
-
- if (writtenSize > reqSize) {
- writtenSize = reqSize;
+ return WAIT_PERIOD_MS * 1000000LL;
}
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// 8 to 16 bit conversion, note that source and destination are the same address
memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
- writtenSize <<= 1;
+ audioBuffer.size <<= 1;
}
- audioBuffer.size = writtenSize;
- // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for
- // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of
- // 16 bit.
- audioBuffer.frameCount = writtenSize / mFrameSizeAF;
-
- frames -= audioBuffer.frameCount;
+ size_t releasedFrames = audioBuffer.size / mFrameSizeAF;
+ audioBuffer.frameCount = releasedFrames;
+ mRemainingFrames -= releasedFrames;
+ if (misalignment >= releasedFrames) {
+ misalignment -= releasedFrames;
+ } else {
+ misalignment = 0;
+ }
releaseBuffer(&audioBuffer);
- }
- while (frames);
- if (frames == 0) {
- mRemainingFrames = mNotificationFramesAct;
- } else {
- mRemainingFrames = frames;
+ // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
+ // if callback doesn't like to accept the full chunk
+ if (writtenSize < reqSize) {
+ continue;
+ }
+
+ // There could be enough non-contiguous frames available to satisfy the remaining request
+ if (mRemainingFrames <= nonContig) {
+ continue;
+ }
+
+#if 0
+ // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
+ // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
+ // that total to a sum == notificationFrames.
+ if (0 < misalignment && misalignment <= mRemainingFrames) {
+ mRemainingFrames = misalignment;
+ return (mRemainingFrames * 1100000000LL) / sampleRate;
+ }
+#endif
+
}
- return true;
+ mRemainingFrames = notificationFrames;
+ mRetryOnPartialBuffer = true;
+
+ // A lot has transpired since ns was calculated, so run again immediately and re-calculate
+ return 0;
}
-// must be called with mLock and refCblk.lock held. Callers must also hold strong references on
-// the IAudioTrack and IMemory in case they are recreated here.
-// If the IAudioTrack is successfully restored, the refCblk pointer is updated
-// FIXME Don't depend on caller to hold strong references.
-status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart)
+status_t AudioTrack::restoreTrack_l(const char *from)
{
+ ALOGW("dead IAudioTrack, creating a new one from %s()", from);
+ ++mSequence;
status_t result;
- audio_track_cblk_t* cblk = refCblk;
- audio_track_cblk_t* newCblk = cblk;
- ALOGW("dead IAudioTrack, creating a new one from %s",
- fromStart ? "start()" : "obtainBuffer()");
-
- // signal old cblk condition so that other threads waiting for available buffers stop
- // waiting now
- cblk->cv.broadcast();
- cblk->lock.unlock();
-
// refresh the audio configuration cache in this process to make sure we get new
// output parameters in getOutput_l() and createTrack_l()
AudioSystem::clearAudioConfigCache();
@@ -1398,68 +1498,47 @@
// if the new IAudioTrack is created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory
+ size_t position = mProxy->getPosition();
+ mNewPosition = position + mUpdatePeriod;
+ size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
result = createTrack_l(mStreamType,
mSampleRate,
mFormat,
mReqFrameCount, // so that frame count never goes down
mFlags,
mSharedBuffer,
- getOutput_l());
+ getOutput_l(),
+ position /*epoch*/);
if (result == NO_ERROR) {
- uint32_t user = cblk->user;
- uint32_t server = cblk->server;
+ // continue playback from last known position, but
+ // don't attempt to restore loop after invalidation; it's difficult and not worthwhile
+ if (mStaticProxy != NULL) {
+ mLoopPeriod = 0;
+ mStaticProxy->setLoop(bufferPosition, mFrameCount, 0);
+ }
+ // FIXME How do we simulate the fact that all frames present in the buffer at the time of
+ // track destruction have been played? This is critical for SoundPool implementation
+ // This must be broken, and needs to be tested/debugged.
+#if 0
// restore write index and set other indexes to reflect empty buffer status
- newCblk = mCblk;
- newCblk->user = user;
- newCblk->server = user;
- newCblk->userBase = user;
- newCblk->serverBase = user;
- // restore loop: this is not guaranteed to succeed if new frame count is not
- // compatible with loop length
- setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
- size_t frames = 0;
- if (!fromStart) {
- newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ if (!strcmp(from, "start")) {
// Make sure that a client relying on callback events indicating underrun or
// the actual amount of audio frames played (e.g SoundPool) receives them.
if (mSharedBuffer == 0) {
- if (user > server) {
- frames = ((user - server) > mFrameCount) ?
- mFrameCount : (user - server);
- memset(mBuffers, 0, frames * mFrameSizeAF);
- }
// restart playback even if buffer is not completely filled.
- android_atomic_or(CBLK_FORCEREADY, &newCblk->flags);
+ android_atomic_or(CBLK_FORCEREADY, &mCblk->flags);
}
}
- if (mSharedBuffer != 0) {
- frames = mFrameCount;
- }
- if (frames > 0) {
- // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to
- // the client
- mProxy->stepUser(frames);
- }
- if (mActive) {
+#endif
+ if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
- ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
- }
- if (fromStart && result == NO_ERROR) {
- mNewPosition = newCblk->server + mUpdatePeriod;
}
}
- ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
- ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
- result, mActive, newCblk, cblk, newCblk->flags, cblk->flags);
-
- if (result == NO_ERROR) {
- // from now on we switch to the newly created cblk
- refCblk = newCblk;
+ if (result != NO_ERROR) {
+ ALOGW("restoreTrack_l() failed status %d", result);
+ mState = STATE_STOPPED;
}
- newCblk->lock.lock();
-
- ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result);
return result;
}
@@ -1480,16 +1559,33 @@
result.append(buffer);
snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus);
result.append(buffer);
- snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
+ snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency);
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
+uint32_t AudioTrack::getUnderrunFrames() const
+{
+ AutoMutex lock(mLock);
+ return mProxy->getUnderrunFrames();
+}
+
+// =========================================================================
+
+void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who)
+{
+ sp<AudioTrack> audioTrack = mAudioTrack.promote();
+ if (audioTrack != 0) {
+ AutoMutex lock(audioTrack->mLock);
+ audioTrack->mProxy->binderDied();
+ }
+}
+
// =========================================================================
AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
- : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
+ : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false)
{
}
@@ -1507,10 +1603,26 @@
return true;
}
}
- if (!mReceiver.processAudioBuffer(this)) {
- pause();
+ nsecs_t ns = mReceiver.processAudioBuffer(this);
+ switch (ns) {
+ case 0:
+ return true;
+ case NS_WHENEVER:
+ sleep(1);
+ return true;
+ case NS_INACTIVE:
+ pauseConditional();
+ return true;
+ case NS_NEVER:
+ return false;
+ default:
+ LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+ struct timespec req;
+ req.tv_sec = ns / 1000000000LL;
+ req.tv_nsec = ns % 1000000000LL;
+ nanosleep(&req, NULL /*rem*/);
+ return true;
}
- return true;
}
void AudioTrack::AudioTrackThread::requestExit()
@@ -1524,6 +1636,17 @@
{
AutoMutex _l(mMyLock);
mPaused = true;
+ mResumeLatch = false;
+}
+
+void AudioTrack::AudioTrackThread::pauseConditional()
+{
+ AutoMutex _l(mMyLock);
+ if (mResumeLatch) {
+ mResumeLatch = false;
+ } else {
+ mPaused = true;
+ }
}
void AudioTrack::AudioTrackThread::resume()
@@ -1531,7 +1654,10 @@
AutoMutex _l(mMyLock);
if (mPaused) {
mPaused = false;
+ mResumeLatch = false;
mMyCond.signal();
+ } else {
+ mResumeLatch = true;
}
}
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 13d47c9..f034164 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -19,178 +19,664 @@
#include <private/media/AudioTrackShared.h>
#include <utils/Log.h>
+extern "C" {
+#include "../private/bionic_futex.h"
+}
namespace android {
audio_track_cblk_t::audio_track_cblk_t()
- : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
- userBase(0), serverBase(0), frameCount_(0),
- loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
- mSampleRate(0), mSendLevel(0), flags(0)
+ : server(0), frameCount_(0), mFutex(0), mMinimum(0),
+ mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mName(0), flags(0)
+{
+ memset(&u, 0, sizeof(u));
+}
+
+// ---------------------------------------------------------------------------
+
+Proxy::Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
+ bool isOut, bool clientInServer)
+ : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize),
+ mFrameCountP2(roundup(frameCount)), mIsOut(isOut), mClientInServer(clientInServer),
+ mIsShutdown(false)
{
}
-uint32_t audio_track_cblk_t::stepUser(size_t stepCount, size_t frameCount, bool isOut)
+// ---------------------------------------------------------------------------
+
+ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize, bool isOut, bool clientInServer)
+ : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer), mEpoch(0)
{
- ALOGV("stepuser %08x %08x %d", user, server, stepCount);
-
- uint32_t u = user;
- u += stepCount;
- // Ensure that user is never ahead of server for AudioRecord
- if (isOut) {
- // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
- if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
- bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
- }
- } else if (u > server) {
- ALOGW("stepUser occurred after track reset");
- u = server;
- }
-
- if (u >= frameCount) {
- // common case, user didn't just wrap
- if (u - frameCount >= userBase ) {
- userBase += frameCount;
- }
- } else if (u >= userBase + frameCount) {
- // user just wrapped
- userBase += frameCount;
- }
-
- user = u;
-
- // Clear flow control error condition as new data has been written/read to/from buffer.
- if (flags & CBLK_UNDERRUN) {
- android_atomic_and(~CBLK_UNDERRUN, &flags);
- }
-
- return u;
}
-bool audio_track_cblk_t::stepServer(size_t stepCount, size_t frameCount, bool isOut)
+const struct timespec ClientProxy::kForever = {INT_MAX /*tv_sec*/, 0 /*tv_nsec*/};
+const struct timespec ClientProxy::kNonBlocking = {0 /*tv_sec*/, 0 /*tv_nsec*/};
+
+#define MEASURE_NS 10000000 // attempt to provide accurate timeouts if requested >= MEASURE_NS
+
+// To facilitate quicker recovery from server failure, this value limits the timeout per each futex
+// wait. However it does not protect infinite timeouts. If defined to be zero, there is no limit.
+// FIXME May not be compatible with audio tunneling requirements where timeout should be in the
+// order of minutes.
+#define MAX_SEC 5
+
+status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *requested,
+ struct timespec *elapsed)
{
- ALOGV("stepserver %08x %08x %d", user, server, stepCount);
-
- if (!tryLock()) {
- ALOGW("stepServer() could not lock cblk");
- return false;
+ if (buffer == NULL || buffer->mFrameCount == 0) {
+ ALOGE("%s BAD_VALUE", __func__);
+ return BAD_VALUE;
}
+ struct timespec total; // total elapsed time spent waiting
+ total.tv_sec = 0;
+ total.tv_nsec = 0;
+ bool measure = elapsed != NULL; // whether to measure total elapsed time spent waiting
- uint32_t s = server;
- bool flushed = (s == user);
-
- s += stepCount;
- if (isOut) {
- // Mark that we have read the first buffer so that next time stepUser() is called
- // we switch to normal obtainBuffer() timeout period
- if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
- bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
- }
- // It is possible that we receive a flush()
- // while the mixer is processing a block: in this case,
- // stepServer() is called After the flush() has reset u & s and
- // we have s > u
- if (flushed) {
- ALOGW("stepServer occurred after track reset");
- s = user;
- }
- }
-
- if (s >= loopEnd) {
- ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
- s = loopStart;
- if (--loopCount == 0) {
- loopEnd = UINT_MAX;
- loopStart = UINT_MAX;
- }
- }
-
- if (s >= frameCount) {
- // common case, server didn't just wrap
- if (s - frameCount >= serverBase ) {
- serverBase += frameCount;
- }
- } else if (s >= serverBase + frameCount) {
- // server just wrapped
- serverBase += frameCount;
- }
-
- server = s;
-
- if (!(flags & CBLK_INVALID)) {
- cv.signal();
- }
- lock.unlock();
- return true;
-}
-
-void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const
-{
- return (int8_t *)buffers + (offset - userBase) * frameSize;
-}
-
-uint32_t audio_track_cblk_t::framesAvailable(size_t frameCount, bool isOut)
-{
- Mutex::Autolock _l(lock);
- return framesAvailable_l(frameCount, isOut);
-}
-
-uint32_t audio_track_cblk_t::framesAvailable_l(size_t frameCount, bool isOut)
-{
- uint32_t u = user;
- uint32_t s = server;
-
- if (isOut) {
- uint32_t limit = (s < loopStart) ? s : loopStart;
- return limit + frameCount - u;
+ status_t status;
+ enum {
+ TIMEOUT_ZERO, // requested == NULL || *requested == 0
+ TIMEOUT_INFINITE, // *requested == infinity
+ TIMEOUT_FINITE, // 0 < *requested < infinity
+ TIMEOUT_CONTINUE, // additional chances after TIMEOUT_FINITE
+ } timeout;
+ if (requested == NULL) {
+ timeout = TIMEOUT_ZERO;
+ } else if (requested->tv_sec == 0 && requested->tv_nsec == 0) {
+ timeout = TIMEOUT_ZERO;
+ } else if (requested->tv_sec == INT_MAX) {
+ timeout = TIMEOUT_INFINITE;
} else {
- return frameCount + u - s;
+ timeout = TIMEOUT_FINITE;
+ if (requested->tv_sec > 0 || requested->tv_nsec >= MEASURE_NS) {
+ measure = true;
+ }
}
-}
-
-uint32_t audio_track_cblk_t::framesReady(bool isOut)
-{
- uint32_t u = user;
- uint32_t s = server;
-
- if (isOut) {
- if (u < loopEnd) {
- return u - s;
+ struct timespec before;
+ bool beforeIsValid = false;
+ audio_track_cblk_t* cblk = mCblk;
+ bool ignoreInitialPendingInterrupt = true;
+ // check for shared memory corruption
+ if (mIsShutdown) {
+ status = NO_INIT;
+ goto end;
+ }
+ for (;;) {
+ int32_t flags = android_atomic_and(~CBLK_INTERRUPT, &cblk->flags);
+ // check for track invalidation by server, or server death detection
+ if (flags & CBLK_INVALID) {
+ ALOGV("Track invalidated");
+ status = DEAD_OBJECT;
+ goto end;
+ }
+ // check for obtainBuffer interrupted by client
+ if (!ignoreInitialPendingInterrupt && (flags & CBLK_INTERRUPT)) {
+ ALOGV("obtainBuffer() interrupted by client");
+ status = -EINTR;
+ goto end;
+ }
+ ignoreInitialPendingInterrupt = false;
+ // compute number of frames available to write (AudioTrack) or read (AudioRecord)
+ int32_t front;
+ int32_t rear;
+ if (mIsOut) {
+ // The barrier following the read of mFront is probably redundant.
+ // We're about to perform a conditional branch based on 'filled',
+ // which will force the processor to observe the read of mFront
+ // prior to allowing data writes starting at mRaw.
+ // However, the processor may support speculative execution,
+ // and be unable to undo speculative writes into shared memory.
+ // The barrier will prevent such speculative execution.
+ front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
+ rear = cblk->u.mStreaming.mRear;
} else {
- // do not block on mutex shared with client on AudioFlinger side
- if (!tryLock()) {
- ALOGW("framesReady() could not lock cblk");
- return 0;
+ // On the other hand, this barrier is required.
+ rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+ front = cblk->u.mStreaming.mFront;
+ }
+ ssize_t filled = rear - front;
+ // pipe should not be overfull
+ if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
+ ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ mIsShutdown = true;
+ status = NO_INIT;
+ goto end;
+ }
+ // don't allow filling pipe beyond the nominal size
+ size_t avail = mIsOut ? mFrameCount - filled : filled;
+ if (avail > 0) {
+ // 'avail' may be non-contiguous, so return only the first contiguous chunk
+ size_t part1;
+ if (mIsOut) {
+ rear &= mFrameCountP2 - 1;
+ part1 = mFrameCountP2 - rear;
+ } else {
+ front &= mFrameCountP2 - 1;
+ part1 = mFrameCountP2 - front;
}
- uint32_t frames = UINT_MAX;
- if (loopCount >= 0) {
- frames = (loopEnd - loopStart)*loopCount + u - s;
+ if (part1 > avail) {
+ part1 = avail;
}
- lock.unlock();
- return frames;
+ if (part1 > buffer->mFrameCount) {
+ part1 = buffer->mFrameCount;
+ }
+ buffer->mFrameCount = part1;
+ buffer->mRaw = part1 > 0 ?
+ &((char *) mBuffers)[(mIsOut ? rear : front) * mFrameSize] : NULL;
+ buffer->mNonContig = avail - part1;
+ // mUnreleased = part1;
+ status = NO_ERROR;
+ break;
+ }
+ struct timespec remaining;
+ const struct timespec *ts;
+ switch (timeout) {
+ case TIMEOUT_ZERO:
+ status = WOULD_BLOCK;
+ goto end;
+ case TIMEOUT_INFINITE:
+ ts = NULL;
+ break;
+ case TIMEOUT_FINITE:
+ timeout = TIMEOUT_CONTINUE;
+ if (MAX_SEC == 0) {
+ ts = requested;
+ break;
+ }
+ // fall through
+ case TIMEOUT_CONTINUE:
+ // FIXME we do not retry if requested < 10ms? needs documentation on this state machine
+ if (!measure || requested->tv_sec < total.tv_sec ||
+ (requested->tv_sec == total.tv_sec && requested->tv_nsec <= total.tv_nsec)) {
+ status = TIMED_OUT;
+ goto end;
+ }
+ remaining.tv_sec = requested->tv_sec - total.tv_sec;
+ if ((remaining.tv_nsec = requested->tv_nsec - total.tv_nsec) < 0) {
+ remaining.tv_nsec += 1000000000;
+ remaining.tv_sec++;
+ }
+ if (0 < MAX_SEC && MAX_SEC < remaining.tv_sec) {
+ remaining.tv_sec = MAX_SEC;
+ remaining.tv_nsec = 0;
+ }
+ ts = &remaining;
+ break;
+ default:
+ LOG_FATAL("%s timeout=%d", timeout);
+ ts = NULL;
+ break;
+ }
+ int32_t old = android_atomic_dec(&cblk->mFutex);
+ if (old <= 0) {
+ int rc;
+ if (measure && !beforeIsValid) {
+ clock_gettime(CLOCK_MONOTONIC, &before);
+ beforeIsValid = true;
+ }
+ int ret = __futex_syscall4(&cblk->mFutex,
+ mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old - 1, ts);
+ // update total elapsed time spent waiting
+ if (measure) {
+ struct timespec after;
+ clock_gettime(CLOCK_MONOTONIC, &after);
+ total.tv_sec += after.tv_sec - before.tv_sec;
+ long deltaNs = after.tv_nsec - before.tv_nsec;
+ if (deltaNs < 0) {
+ deltaNs += 1000000000;
+ total.tv_sec--;
+ }
+ if ((total.tv_nsec += deltaNs) >= 1000000000) {
+ total.tv_nsec -= 1000000000;
+ total.tv_sec++;
+ }
+ before = after;
+ beforeIsValid = true;
+ }
+ switch (ret) {
+ case 0: // normal wakeup by server, or by binderDied()
+ case -EWOULDBLOCK: // benign race condition with server
+ case -EINTR: // wait was interrupted by signal or other spurious wakeup
+ case -ETIMEDOUT: // time-out expired
+ break;
+ default:
+ ALOGE("%s unexpected error %d", __func__, ret);
+ status = -ret;
+ goto end;
+ }
+ }
+ }
+
+end:
+ if (status != NO_ERROR) {
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ }
+ if (elapsed != NULL) {
+ *elapsed = total;
+ }
+ if (requested == NULL) {
+ requested = &kNonBlocking;
+ }
+ if (measure) {
+ ALOGV("requested %d.%03d elapsed %d.%03d", requested->tv_sec, requested->tv_nsec / 1000000,
+ total.tv_sec, total.tv_nsec / 1000000);
+ }
+ return status;
+}
+
+void ClientProxy::releaseBuffer(Buffer* buffer)
+{
+ size_t stepCount = buffer->mFrameCount;
+ // FIXME
+ // check mUnreleased
+ // verify that stepCount <= frameCount returned by the last obtainBuffer()
+ // verify stepCount not > total frame count of pipe
+ if (stepCount == 0) {
+ return;
+ }
+ audio_track_cblk_t* cblk = mCblk;
+ // Both of these barriers are required
+ if (mIsOut) {
+ int32_t rear = cblk->u.mStreaming.mRear;
+ android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear);
+ } else {
+ int32_t front = cblk->u.mStreaming.mFront;
+ android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront);
+ }
+}
+
+void ClientProxy::binderDied()
+{
+ audio_track_cblk_t* cblk = mCblk;
+ if (!(android_atomic_or(CBLK_INVALID, &cblk->flags) & CBLK_INVALID)) {
+ // it seems that a FUTEX_WAKE_PRIVATE will not wake a FUTEX_WAIT, even within same process
+ (void) __futex_syscall3(&cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE,
+ 1);
+ }
+}
+
+void ClientProxy::interrupt()
+{
+ audio_track_cblk_t* cblk = mCblk;
+ if (!(android_atomic_or(CBLK_INTERRUPT, &cblk->flags) & CBLK_INTERRUPT)) {
+ (void) __futex_syscall3(&cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE,
+ 1);
+ }
+}
+
+size_t ClientProxy::getMisalignment()
+{
+ audio_track_cblk_t* cblk = mCblk;
+ return (mFrameCountP2 - (mIsOut ? cblk->u.mStreaming.mRear : cblk->u.mStreaming.mFront)) &
+ (mFrameCountP2 - 1);
+}
+
+// ---------------------------------------------------------------------------
+
+void AudioTrackClientProxy::flush()
+{
+ mCblk->u.mStreaming.mFlush++;
+}
+
+// ---------------------------------------------------------------------------
+
+StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers,
+ size_t frameCount, size_t frameSize)
+ : AudioTrackClientProxy(cblk, buffers, frameCount, frameSize),
+ mMutator(&cblk->u.mStatic.mSingleStateQueue), mBufferPosition(0)
+{
+}
+
+void StaticAudioTrackClientProxy::flush()
+{
+ LOG_FATAL("static flush");
+}
+
+void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
+{
+ StaticAudioTrackState newState;
+ newState.mLoopStart = loopStart;
+ newState.mLoopEnd = loopEnd;
+ newState.mLoopCount = loopCount;
+ mBufferPosition = loopStart;
+ (void) mMutator.push(newState);
+}
+
+size_t StaticAudioTrackClientProxy::getBufferPosition()
+{
+ size_t bufferPosition;
+ if (mMutator.ack()) {
+ bufferPosition = mCblk->u.mStatic.mBufferPosition;
+ if (bufferPosition > mFrameCount) {
+ bufferPosition = mFrameCount;
}
} else {
- return s - u;
+ bufferPosition = mBufferPosition;
}
+ return bufferPosition;
}
-bool audio_track_cblk_t::tryLock()
+// ---------------------------------------------------------------------------
+
+ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
+ size_t frameSize, bool isOut, bool clientInServer)
+ : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer), mUnreleased(0),
+ mAvailToClient(0), mFlush(0), mDeferWake(false)
{
- // the code below simulates lock-with-timeout
- // we MUST do this to protect the AudioFlinger server
- // as this lock is shared with the client.
- status_t err;
-
- err = lock.tryLock();
- if (err == -EBUSY) { // just wait a bit
- usleep(1000);
- err = lock.tryLock();
- }
- if (err != NO_ERROR) {
- // probably, the client just died.
- return false;
- }
- return true;
}
+status_t ServerProxy::obtainBuffer(Buffer* buffer)
+{
+ if (mIsShutdown) {
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ mUnreleased = 0;
+ return NO_INIT;
+ }
+ audio_track_cblk_t* cblk = mCblk;
+ // compute number of frames available to write (AudioTrack) or read (AudioRecord),
+ // or use previous cached value from framesReady(), with added barrier if it omits.
+ int32_t front;
+ int32_t rear;
+ // See notes on barriers at ClientProxy::obtainBuffer()
+ if (mIsOut) {
+ int32_t flush = cblk->u.mStreaming.mFlush;
+ rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+ if (flush != mFlush) {
+ front = rear;
+ mFlush = flush;
+ } else {
+ front = cblk->u.mStreaming.mFront;
+ }
+ } else {
+ front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
+ rear = cblk->u.mStreaming.mRear;
+ }
+ ssize_t filled = rear - front;
+ // pipe should not already be overfull
+ if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
+ ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ mIsShutdown = true;
+ }
+ if (mIsShutdown) {
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ mUnreleased = 0;
+ return NO_INIT;
+ }
+ // don't allow filling pipe beyond the nominal size
+ size_t availToServer;
+ if (mIsOut) {
+ availToServer = filled;
+ mAvailToClient = mFrameCount - filled;
+ } else {
+ availToServer = mFrameCount - filled;
+ mAvailToClient = filled;
+ }
+ // 'availToServer' may be non-contiguous, so return only the first contiguous chunk
+ size_t part1;
+ if (mIsOut) {
+ front &= mFrameCountP2 - 1;
+ part1 = mFrameCountP2 - front;
+ } else {
+ rear &= mFrameCountP2 - 1;
+ part1 = mFrameCountP2 - rear;
+ }
+ if (part1 > availToServer) {
+ part1 = availToServer;
+ }
+ size_t ask = buffer->mFrameCount;
+ if (part1 > ask) {
+ part1 = ask;
+ }
+ // is assignment redundant in some cases?
+ buffer->mFrameCount = part1;
+ buffer->mRaw = part1 > 0 ?
+ &((char *) mBuffers)[(mIsOut ? front : rear) * mFrameSize] : NULL;
+ buffer->mNonContig = availToServer - part1;
+ mUnreleased = part1;
+ // optimization to avoid waking up the client too early
+ // FIXME need to test for recording
+ mDeferWake = part1 < ask && availToServer >= ask;
+ return part1 > 0 ? NO_ERROR : WOULD_BLOCK;
+}
+
+void ServerProxy::releaseBuffer(Buffer* buffer)
+{
+ if (mIsShutdown) {
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ return;
+ }
+ size_t stepCount = buffer->mFrameCount;
+ LOG_ALWAYS_FATAL_IF(stepCount > mUnreleased);
+ if (stepCount == 0) {
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ return;
+ }
+ mUnreleased -= stepCount;
+ audio_track_cblk_t* cblk = mCblk;
+ if (mIsOut) {
+ int32_t front = cblk->u.mStreaming.mFront;
+ android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront);
+ } else {
+ int32_t rear = cblk->u.mStreaming.mRear;
+ android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear);
+ }
+
+ mCblk->server += stepCount;
+
+ size_t half = mFrameCount / 2;
+ if (half == 0) {
+ half = 1;
+ }
+ size_t minimum = cblk->mMinimum;
+ if (minimum == 0) {
+ minimum = mIsOut ? half : 1;
+ } else if (minimum > half) {
+ minimum = half;
+ }
+ if (!mDeferWake && mAvailToClient + stepCount >= minimum) {
+ ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum);
+ // could client be sleeping, or not need this increment and counter overflows?
+ int32_t old = android_atomic_inc(&cblk->mFutex);
+ if (old == -1) {
+ (void) __futex_syscall3(&cblk->mFutex,
+ mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1);
+ }
+ }
+
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+}
+
+// ---------------------------------------------------------------------------
+
+size_t AudioTrackServerProxy::framesReady()
+{
+ LOG_ALWAYS_FATAL_IF(!mIsOut);
+
+ if (mIsShutdown) {
+ return 0;
+ }
+ audio_track_cblk_t* cblk = mCblk;
+ // the acquire might not be necessary since not doing a subsequent read
+ int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+ ssize_t filled = rear - cblk->u.mStreaming.mFront;
+ // pipe should not already be overfull
+ if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
+ ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ mIsShutdown = true;
+ return 0;
+ }
+ // cache this value for later use by obtainBuffer(), with added barrier
+ // and racy if called by normal mixer thread
+ // ignores flush(), so framesReady() may report a larger mFrameCount than obtainBuffer()
+ return filled;
+}
+
+// ---------------------------------------------------------------------------
+
+StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers,
+ size_t frameCount, size_t frameSize)
+ : AudioTrackServerProxy(cblk, buffers, frameCount, frameSize),
+ mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0),
+ mEnd(frameCount), mFramesReadyIsCalledByMultipleThreads(false)
+{
+ mState.mLoopStart = 0;
+ mState.mLoopEnd = 0;
+ mState.mLoopCount = 0;
+}
+
+void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads()
+{
+ mFramesReadyIsCalledByMultipleThreads = true;
+}
+
+size_t StaticAudioTrackServerProxy::framesReady()
+{
+ // FIXME
+ // This is racy if called by normal mixer thread,
+ // as we're reading 2 independent variables without a lock.
+ // Can't call mObserver.poll(), as we might be called from wrong thread.
+ // If looping is enabled, should return a higher number (since includes non-contiguous).
+ size_t position = mPosition;
+ if (!mFramesReadyIsCalledByMultipleThreads) {
+ ssize_t positionOrStatus = pollPosition();
+ if (positionOrStatus >= 0) {
+ position = (size_t) positionOrStatus;
+ }
+ }
+ size_t end = mEnd;
+ return position < end ? end - position : 0;
+}
+
+ssize_t StaticAudioTrackServerProxy::pollPosition()
+{
+ size_t position = mPosition;
+ StaticAudioTrackState state;
+ if (mObserver.poll(state)) {
+ bool valid = false;
+ size_t loopStart = state.mLoopStart;
+ size_t loopEnd = state.mLoopEnd;
+ if (state.mLoopCount == 0) {
+ if (loopStart > mFrameCount) {
+ loopStart = mFrameCount;
+ }
+ // ignore loopEnd
+ mPosition = position = loopStart;
+ mEnd = mFrameCount;
+ mState.mLoopCount = 0;
+ valid = true;
+ } else {
+ if (loopStart < loopEnd && loopEnd <= mFrameCount &&
+ loopEnd - loopStart >= MIN_LOOP) {
+ if (!(loopStart <= position && position < loopEnd)) {
+ mPosition = position = loopStart;
+ }
+ mEnd = loopEnd;
+ mState = state;
+ valid = true;
+ }
+ }
+ if (!valid) {
+ ALOGE("%s client pushed an invalid state, shutting down", __func__);
+ mIsShutdown = true;
+ return (ssize_t) NO_INIT;
+ }
+ mCblk->u.mStatic.mBufferPosition = position;
+ }
+ return (ssize_t) position;
+}
+
+status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer)
+{
+ if (mIsShutdown) {
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ mUnreleased = 0;
+ return NO_INIT;
+ }
+ ssize_t positionOrStatus = pollPosition();
+ if (positionOrStatus < 0) {
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ mUnreleased = 0;
+ return (status_t) positionOrStatus;
+ }
+ size_t position = (size_t) positionOrStatus;
+ size_t avail;
+ if (position < mEnd) {
+ avail = mEnd - position;
+ size_t wanted = buffer->mFrameCount;
+ if (avail < wanted) {
+ buffer->mFrameCount = avail;
+ } else {
+ avail = wanted;
+ }
+ buffer->mRaw = &((char *) mBuffers)[position * mFrameSize];
+ } else {
+ avail = 0;
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ }
+ buffer->mNonContig = 0; // FIXME should be > 0 for looping
+ mUnreleased = avail;
+ return NO_ERROR;
+}
+
+void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
+{
+ size_t stepCount = buffer->mFrameCount;
+ LOG_ALWAYS_FATAL_IF(stepCount > mUnreleased);
+ if (stepCount == 0) {
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+ return;
+ }
+ mUnreleased -= stepCount;
+ audio_track_cblk_t* cblk = mCblk;
+ size_t position = mPosition;
+ size_t newPosition = position + stepCount;
+ int32_t setFlags = 0;
+ if (!(position <= newPosition && newPosition <= mFrameCount)) {
+ ALOGW("%s newPosition %u outside [%u, %u]", __func__, newPosition, position, mFrameCount);
+ newPosition = mFrameCount;
+ } else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
+ if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) {
+ newPosition = mState.mLoopStart;
+ setFlags = CBLK_LOOP_CYCLE;
+ } else {
+ mEnd = mFrameCount; // this is what allows playback to continue after the loop
+ setFlags = CBLK_LOOP_FINAL;
+ }
+ }
+ if (newPosition == mFrameCount) {
+ setFlags |= CBLK_BUFFER_END;
+ }
+ mPosition = newPosition;
+
+ cblk->server += stepCount;
+ cblk->u.mStatic.mBufferPosition = newPosition;
+ if (setFlags != 0) {
+ (void) android_atomic_or(setFlags, &cblk->flags);
+ // this would be a good place to wake a futex
+ }
+
+ buffer->mFrameCount = 0;
+ buffer->mRaw = NULL;
+ buffer->mNonContig = 0;
+}
+
+// ---------------------------------------------------------------------------
+
} // namespace android
diff --git a/media/libmedia/IHDCP.cpp b/media/libmedia/IHDCP.cpp
index f13addc..a46ff91 100644
--- a/media/libmedia/IHDCP.cpp
+++ b/media/libmedia/IHDCP.cpp
@@ -31,6 +31,7 @@
HDCP_INIT_ASYNC,
HDCP_SHUTDOWN_ASYNC,
HDCP_ENCRYPT,
+ HDCP_ENCRYPT_NATIVE,
HDCP_DECRYPT,
};
@@ -108,6 +109,31 @@
return err;
}
+ virtual status_t encryptNative(
+ const sp<GraphicBuffer> &graphicBuffer,
+ size_t offset, size_t size, uint32_t streamCTR,
+ uint64_t *outInputCTR, void *outData) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IHDCP::getInterfaceDescriptor());
+ data.write(*graphicBuffer);
+ data.writeInt32(offset);
+ data.writeInt32(size);
+ data.writeInt32(streamCTR);
+ remote()->transact(HDCP_ENCRYPT_NATIVE, data, &reply);
+
+ status_t err = reply.readInt32();
+
+ if (err != OK) {
+ *outInputCTR = 0;
+ return err;
+ }
+
+ *outInputCTR = reply.readInt64();
+ reply.read(outData, size);
+
+ return err;
+ }
+
virtual status_t decrypt(
const void *inData, size_t size,
uint32_t streamCTR, uint64_t inputCTR,
@@ -222,6 +248,34 @@
return OK;
}
+ case HDCP_ENCRYPT_NATIVE:
+ {
+ CHECK_INTERFACE(IHDCP, data, reply);
+
+ sp<GraphicBuffer> graphicBuffer = new GraphicBuffer();
+ data.read(*graphicBuffer);
+ size_t offset = data.readInt32();
+ size_t size = data.readInt32();
+ uint32_t streamCTR = data.readInt32();
+ void *outData = malloc(size);
+ uint64_t inputCTR;
+
+ status_t err = encryptNative(graphicBuffer, offset, size,
+ streamCTR, &inputCTR, outData);
+
+ reply->writeInt32(err);
+
+ if (err == OK) {
+ reply->writeInt64(inputCTR);
+ reply->write(outData, size);
+ }
+
+ free(outData);
+ outData = NULL;
+
+ return OK;
+ }
+
case HDCP_DECRYPT:
{
size_t size = data.readInt32();
diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 9199db6..9db5b1b 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -49,10 +49,10 @@
} while (true);
if (sDeathNotifier == NULL) {
- sDeathNotifier = new DeathNotifier();
- }
- binder->linkToDeath(sDeathNotifier);
- sMediaPlayerService = interface_cast<IMediaPlayerService>(binder);
+ sDeathNotifier = new DeathNotifier();
+ }
+ binder->linkToDeath(sDeathNotifier);
+ sMediaPlayerService = interface_cast<IMediaPlayerService>(binder);
}
ALOGE_IF(sMediaPlayerService == 0, "no media player service!?");
return sMediaPlayerService;
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 59e538f..8fe5bb3 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -39,7 +39,6 @@
mMaxTracks(maxTracks),
mEasData(NULL),
mEasJetFileLoc(NULL),
- mAudioTrack(NULL),
mTrackBufferSize(trackBufferSize)
{
ALOGV("JetPlayer constructor");
@@ -140,11 +139,10 @@
free(mEasJetFileLoc);
mEasJetFileLoc = NULL;
}
- if (mAudioTrack) {
+ if (mAudioTrack != 0) {
mAudioTrack->stop();
mAudioTrack->flush();
- delete mAudioTrack;
- mAudioTrack = NULL;
+ mAudioTrack.clear();
}
if (mAudioBuffer) {
delete mAudioBuffer;
diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index e1e3348..93a4a4c 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -16,7 +16,7 @@
#include <media/mediascanner.h>
-#include <utils/StringArray.h>
+#include "StringArray.h"
#include "autodetect.h"
#include "unicode/ucnv.h"
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index ee70ef7..e1e88ec 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -547,8 +547,8 @@
void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume,
float rightVolume, int priority, int loop, float rate)
{
- AudioTrack* oldTrack;
- AudioTrack* newTrack;
+ sp<AudioTrack> oldTrack;
+ sp<AudioTrack> newTrack;
status_t status;
{ // scope for the lock
@@ -620,7 +620,7 @@
ALOGE("Error creating AudioTrack");
goto exit;
}
- ALOGV("setVolume %p", newTrack);
+ ALOGV("setVolume %p", newTrack.get());
newTrack->setVolume(leftVolume, rightVolume);
newTrack->setLoop(0, frameCount, loop);
@@ -643,11 +643,9 @@
}
exit:
- ALOGV("delete oldTrack %p", oldTrack);
- delete oldTrack;
+ ALOGV("delete oldTrack %p", oldTrack.get());
if (status != NO_ERROR) {
- delete newTrack;
- mAudioTrack = NULL;
+ mAudioTrack.clear();
}
}
@@ -884,7 +882,7 @@
}
// do not call AudioTrack destructor with mLock held as it will wait for the AudioTrack
// callback thread to exit which may need to execute process() and acquire the mLock.
- delete mAudioTrack;
+ mAudioTrack.clear();
}
void SoundChannel::dump()
diff --git a/media/libmedia/StringArray.cpp b/media/libmedia/StringArray.cpp
new file mode 100644
index 0000000..5f5b57a
--- /dev/null
+++ b/media/libmedia/StringArray.cpp
@@ -0,0 +1,113 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//
+// Sortable array of strings. STL-ish, but STL-free.
+//
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "StringArray.h"
+
+namespace android {
+
+//
+// An expanding array of strings. Add, get, sort, delete.
+//
+StringArray::StringArray()
+ : mMax(0), mCurrent(0), mArray(NULL)
+{
+}
+
+StringArray:: ~StringArray() {
+ for (int i = 0; i < mCurrent; i++)
+ delete[] mArray[i];
+ delete[] mArray;
+}
+
+//
+// Add a string. A copy of the string is made.
+//
+bool StringArray::push_back(const char* str) {
+ if (mCurrent >= mMax) {
+ char** tmp;
+
+ if (mMax == 0)
+ mMax = 16; // initial storage
+ else
+ mMax *= 2;
+
+ tmp = new char*[mMax];
+ if (tmp == NULL)
+ return false;
+
+ memcpy(tmp, mArray, mCurrent * sizeof(char*));
+ delete[] mArray;
+ mArray = tmp;
+ }
+
+ int len = strlen(str);
+ mArray[mCurrent] = new char[len+1];
+ memcpy(mArray[mCurrent], str, len+1);
+ mCurrent++;
+
+ return true;
+}
+
+//
+// Delete an entry.
+//
+void StringArray::erase(int idx) {
+ if (idx < 0 || idx >= mCurrent)
+ return;
+ delete[] mArray[idx];
+ if (idx < mCurrent-1) {
+ memmove(&mArray[idx], &mArray[idx+1],
+ (mCurrent-1 - idx) * sizeof(char*));
+ }
+ mCurrent--;
+}
+
+//
+// Sort the array.
+//
+void StringArray::sort(int (*compare)(const void*, const void*)) {
+ qsort(mArray, mCurrent, sizeof(char*), compare);
+}
+
+//
+// Pass this to the sort routine to do an ascending alphabetical sort.
+//
+int StringArray::cmpAscendingAlpha(const void* pstr1, const void* pstr2) {
+ return strcmp(*(const char**)pstr1, *(const char**)pstr2);
+}
+
+//
+// Set entry N to specified string.
+// [should use operator[] here]
+//
+void StringArray::setEntry(int idx, const char* str) {
+ if (idx < 0 || idx >= mCurrent)
+ return;
+ delete[] mArray[idx];
+ int len = strlen(str);
+ mArray[idx] = new char[len+1];
+ memcpy(mArray[idx], str, len+1);
+}
+
+
+}; // namespace android
diff --git a/media/libmedia/StringArray.h b/media/libmedia/StringArray.h
new file mode 100644
index 0000000..ae47085
--- /dev/null
+++ b/media/libmedia/StringArray.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//
+// Sortable array of strings. STL-ish, but STL-free.
+//
+#ifndef _LIBS_MEDIA_STRING_ARRAY_H
+#define _LIBS_MEDIA_STRING_ARRAY_H
+
+#include <stdlib.h>
+#include <string.h>
+
+namespace android {
+
+//
+// An expanding array of strings. Add, get, sort, delete.
+//
+class StringArray {
+public:
+ StringArray();
+ virtual ~StringArray();
+
+ //
+ // Add a string. A copy of the string is made.
+ //
+ bool push_back(const char* str);
+
+ //
+ // Delete an entry.
+ //
+ void erase(int idx);
+
+ //
+ // Sort the array.
+ //
+ void sort(int (*compare)(const void*, const void*));
+
+ //
+ // Pass this to the sort routine to do an ascending alphabetical sort.
+ //
+ static int cmpAscendingAlpha(const void* pstr1, const void* pstr2);
+
+ //
+ // Get the #of items in the array.
+ //
+ inline int size(void) const { return mCurrent; }
+
+ //
+ // Return entry N.
+ // [should use operator[] here]
+ //
+ const char* getEntry(int idx) const {
+ return (unsigned(idx) >= unsigned(mCurrent)) ? NULL : mArray[idx];
+ }
+
+ //
+ // Set entry N to specified string.
+ // [should use operator[] here]
+ //
+ void setEntry(int idx, const char* str);
+
+private:
+ int mMax;
+ int mCurrent;
+ char** mArray;
+};
+
+}; // namespace android
+
+#endif // _LIBS_MEDIA_STRING_ARRAY_H
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index f55b697..f9ad31d 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -803,7 +803,6 @@
ALOGV("ToneGenerator constructor: streamType=%d, volume=%f", streamType, volume);
mState = TONE_IDLE;
- mpAudioTrack = NULL;
if (AudioSystem::getOutputSamplingRate(&mSamplingRate, streamType) != NO_ERROR) {
ALOGE("Unable to marshal AudioFlinger");
@@ -855,10 +854,10 @@
ToneGenerator::~ToneGenerator() {
ALOGV("ToneGenerator destructor");
- if (mpAudioTrack != NULL) {
+ if (mpAudioTrack != 0) {
stopTone();
- ALOGV("Delete Track: %p", mpAudioTrack);
- delete mpAudioTrack;
+ ALOGV("Delete Track: %p", mpAudioTrack.get());
+ mpAudioTrack.clear();
}
}
@@ -1047,14 +1046,9 @@
////////////////////////////////////////////////////////////////////////////////
bool ToneGenerator::initAudioTrack() {
- if (mpAudioTrack) {
- delete mpAudioTrack;
- mpAudioTrack = NULL;
- }
-
// Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
mpAudioTrack = new AudioTrack();
- ALOGV("Create Track: %p", mpAudioTrack);
+ ALOGV("Create Track: %p", mpAudioTrack.get());
mpAudioTrack->set(mStreamType,
0, // sampleRate
@@ -1066,7 +1060,9 @@
this, // user
0, // notificationFrames
0, // sharedBuffer
- mThreadCanCallJava);
+ mThreadCanCallJava,
+ 0, // sessionId
+ AudioTrack::TRANSFER_CALLBACK);
if (mpAudioTrack->initCheck() != NO_ERROR) {
ALOGE("AudioTrack->initCheck failed");
@@ -1081,12 +1077,10 @@
initAudioTrack_exit:
+ ALOGV("Init failed: %p", mpAudioTrack.get());
+
// Cleanup
- if (mpAudioTrack != NULL) {
- ALOGV("Delete Track I: %p", mpAudioTrack);
- delete mpAudioTrack;
- mpAudioTrack = NULL;
- }
+ mpAudioTrack.clear();
return false;
}
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index d87bc7f..8f21632 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -34,6 +34,7 @@
libsonivox \
libstagefright \
libstagefright_foundation \
+ libstagefright_httplive \
libstagefright_omx \
libstagefright_wfd \
libutils \
diff --git a/media/libmediaplayerservice/HDCP.cpp b/media/libmediaplayerservice/HDCP.cpp
index 469a02e..8a3188c 100644
--- a/media/libmediaplayerservice/HDCP.cpp
+++ b/media/libmediaplayerservice/HDCP.cpp
@@ -116,6 +116,24 @@
return mHDCPModule->encrypt(inData, size, streamCTR, outInputCTR, outData);
}
+status_t HDCP::encryptNative(
+ const sp<GraphicBuffer> &graphicBuffer,
+ size_t offset, size_t size, uint32_t streamCTR,
+ uint64_t *outInputCTR, void *outData) {
+ Mutex::Autolock autoLock(mLock);
+
+ CHECK(mIsEncryptionModule);
+
+ if (mHDCPModule == NULL) {
+ *outInputCTR = 0;
+
+ return NO_INIT;
+ }
+
+ return mHDCPModule->encryptNative(graphicBuffer->handle,
+ offset, size, streamCTR, outInputCTR, outData);
+}
+
status_t HDCP::decrypt(
const void *inData, size_t size,
uint32_t streamCTR, uint64_t outInputCTR, void *outData) {
diff --git a/media/libmediaplayerservice/HDCP.h b/media/libmediaplayerservice/HDCP.h
index 42e6467..c60c2e0 100644
--- a/media/libmediaplayerservice/HDCP.h
+++ b/media/libmediaplayerservice/HDCP.h
@@ -35,6 +35,11 @@
const void *inData, size_t size, uint32_t streamCTR,
uint64_t *outInputCTR, void *outData);
+ virtual status_t encryptNative(
+ const sp<GraphicBuffer> &graphicBuffer,
+ size_t offset, size_t size, uint32_t streamCTR,
+ uint64_t *outInputCTR, void *outData);
+
virtual status_t decrypt(
const void *inData, size_t size,
uint32_t streamCTR, uint64_t outInputCTR, void *outData);
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index e600a3f..fa1ff36 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1295,8 +1295,6 @@
mSessionId(sessionId),
mFlags(AUDIO_OUTPUT_FLAG_NONE) {
ALOGV("AudioOutput(%d)", sessionId);
- mTrack = 0;
- mRecycledTrack = 0;
mStreamType = AUDIO_STREAM_MUSIC;
mLeftVolume = 1.0;
mRightVolume = 1.0;
@@ -1311,7 +1309,6 @@
MediaPlayerService::AudioOutput::~AudioOutput()
{
close();
- delete mRecycledTrack;
delete mCallbackData;
}
@@ -1422,7 +1419,7 @@
}
}
- AudioTrack *t;
+ sp<AudioTrack> t;
CallbackData *newcbd = NULL;
if (mCallback != NULL) {
newcbd = new CallbackData(this);
@@ -1453,13 +1450,12 @@
if ((t == 0) || (t->initCheck() != NO_ERROR)) {
ALOGE("Unable to create audio track");
- delete t;
delete newcbd;
return NO_INIT;
}
- if (mRecycledTrack) {
+ if (mRecycledTrack != 0) {
// check if the existing track can be reused as-is, or if a new track needs to be created.
bool reuse = true;
@@ -1484,11 +1480,10 @@
ALOGV("chaining to next output");
close();
mTrack = mRecycledTrack;
- mRecycledTrack = NULL;
+ mRecycledTrack.clear();
if (mCallbackData != NULL) {
mCallbackData->setOutput(this);
}
- delete t;
delete newcbd;
return OK;
}
@@ -1499,8 +1494,7 @@
mCallbackData->endTrackSwitch();
}
mRecycledTrack->flush();
- delete mRecycledTrack;
- mRecycledTrack = NULL;
+ mRecycledTrack.clear();
delete mCallbackData;
mCallbackData = NULL;
close();
@@ -1533,7 +1527,7 @@
if (mCallbackData != NULL) {
mCallbackData->endTrackSwitch();
}
- if (mTrack) {
+ if (mTrack != 0) {
mTrack->setVolume(mLeftVolume, mRightVolume);
mTrack->setAuxEffectSendLevel(mSendLevel);
mTrack->start();
@@ -1555,7 +1549,7 @@
mNextOutput->mCallbackData = mCallbackData;
mCallbackData = NULL;
mNextOutput->mRecycledTrack = mTrack;
- mTrack = NULL;
+ mTrack.clear();
mNextOutput->mSampleRateHz = mSampleRateHz;
mNextOutput->mMsecsPerFrame = mMsecsPerFrame;
mNextOutput->mBytesWritten = mBytesWritten;
@@ -1568,7 +1562,7 @@
LOG_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
//ALOGV("write(%p, %u)", buffer, size);
- if (mTrack) {
+ if (mTrack != 0) {
ssize_t ret = mTrack->write(buffer, size);
mBytesWritten += ret;
return ret;
@@ -1579,26 +1573,25 @@
void MediaPlayerService::AudioOutput::stop()
{
ALOGV("stop");
- if (mTrack) mTrack->stop();
+ if (mTrack != 0) mTrack->stop();
}
void MediaPlayerService::AudioOutput::flush()
{
ALOGV("flush");
- if (mTrack) mTrack->flush();
+ if (mTrack != 0) mTrack->flush();
}
void MediaPlayerService::AudioOutput::pause()
{
ALOGV("pause");
- if (mTrack) mTrack->pause();
+ if (mTrack != 0) mTrack->pause();
}
void MediaPlayerService::AudioOutput::close()
{
ALOGV("close");
- delete mTrack;
- mTrack = 0;
+ mTrack.clear();
}
void MediaPlayerService::AudioOutput::setVolume(float left, float right)
@@ -1606,7 +1599,7 @@
ALOGV("setVolume(%f, %f)", left, right);
mLeftVolume = left;
mRightVolume = right;
- if (mTrack) {
+ if (mTrack != 0) {
mTrack->setVolume(left, right);
}
}
@@ -1615,7 +1608,7 @@
{
ALOGV("setPlaybackRatePermille(%d)", ratePermille);
status_t res = NO_ERROR;
- if (mTrack) {
+ if (mTrack != 0) {
res = mTrack->setSampleRate(ratePermille * mSampleRateHz / 1000);
} else {
res = NO_INIT;
@@ -1631,7 +1624,7 @@
{
ALOGV("setAuxEffectSendLevel(%f)", level);
mSendLevel = level;
- if (mTrack) {
+ if (mTrack != 0) {
return mTrack->setAuxEffectSendLevel(level);
}
return NO_ERROR;
@@ -1641,7 +1634,7 @@
{
ALOGV("attachAuxEffect(%d)", effectId);
mAuxEffectId = effectId;
- if (mTrack) {
+ if (mTrack != 0) {
return mTrack->attachAuxEffect(effectId);
}
return NO_ERROR;
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index b33805d..e586156 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -78,7 +78,7 @@
AudioOutput(int sessionId);
virtual ~AudioOutput();
- virtual bool ready() const { return mTrack != NULL; }
+ virtual bool ready() const { return mTrack != 0; }
virtual bool realtime() const { return true; }
virtual ssize_t bufferSize() const;
virtual ssize_t frameCount() const;
@@ -120,8 +120,8 @@
static void CallbackWrapper(
int event, void *me, void *info);
- AudioTrack* mTrack;
- AudioTrack* mRecycledTrack;
+ sp<AudioTrack> mTrack;
+ sp<AudioTrack> mRecycledTrack;
sp<AudioOutput> mNextOutput;
AudioCallback mCallback;
void * mCallbackCookie;
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
index 655ee55..c8901ce 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
@@ -20,7 +20,6 @@
#include "HTTPLiveSource.h"
-#include "ATSParser.h"
#include "AnotherPacketSource.h"
#include "LiveDataSource.h"
#include "LiveSession.h"
@@ -62,7 +61,10 @@
NuPlayer::HTTPLiveSource::~HTTPLiveSource() {
if (mLiveSession != NULL) {
mLiveSession->disconnect();
+ mLiveSession.clear();
+
mLiveLooper->stop();
+ mLiveLooper.clear();
}
}
@@ -76,112 +78,42 @@
mLiveSession = new LiveSession(
notify,
(mFlags & kFlagIncognito) ? LiveSession::kFlagIncognito : 0,
- mUIDValid, mUID);
+ mUIDValid,
+ mUID);
mLiveLooper->registerHandler(mLiveSession);
- mLiveSession->connect(
+ mLiveSession->connectAsync(
mURL.c_str(), mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
-
- mTSParser = new ATSParser;
}
void NuPlayer::HTTPLiveSource::start() {
}
-sp<MetaData> NuPlayer::HTTPLiveSource::getFormatMeta(bool audio) {
- ATSParser::SourceType type =
- audio ? ATSParser::AUDIO : ATSParser::VIDEO;
+sp<AMessage> NuPlayer::HTTPLiveSource::getFormat(bool audio) {
+ sp<AMessage> format;
+ status_t err = mLiveSession->getStreamFormat(
+ audio ? LiveSession::STREAMTYPE_AUDIO
+ : LiveSession::STREAMTYPE_VIDEO,
+ &format);
- sp<AnotherPacketSource> source =
- static_cast<AnotherPacketSource *>(mTSParser->getSource(type).get());
-
- if (source == NULL) {
+ if (err != OK) {
return NULL;
}
- return source->getFormat();
+ return format;
}
status_t NuPlayer::HTTPLiveSource::feedMoreTSData() {
- if (mFinalResult != OK) {
- return mFinalResult;
- }
-
- sp<LiveDataSource> source =
- static_cast<LiveDataSource *>(mLiveSession->getDataSource().get());
-
- for (int32_t i = 0; i < 50; ++i) {
- char buffer[188];
- ssize_t n = source->readAtNonBlocking(mOffset, buffer, sizeof(buffer));
-
- if (n == -EWOULDBLOCK) {
- break;
- } else if (n < 0) {
- if (n != ERROR_END_OF_STREAM) {
- ALOGI("input data EOS reached, error %ld", n);
- } else {
- ALOGI("input data EOS reached.");
- }
- mTSParser->signalEOS(n);
- mFinalResult = n;
- break;
- } else {
- if (buffer[0] == 0x00) {
- // XXX legacy
-
- uint8_t type = buffer[1];
-
- sp<AMessage> extra = new AMessage;
-
- if (type & 2) {
- int64_t mediaTimeUs;
- memcpy(&mediaTimeUs, &buffer[2], sizeof(mediaTimeUs));
-
- extra->setInt64(IStreamListener::kKeyMediaTimeUs, mediaTimeUs);
- }
-
- mTSParser->signalDiscontinuity(
- ((type & 1) == 0)
- ? ATSParser::DISCONTINUITY_SEEK
- : ATSParser::DISCONTINUITY_FORMATCHANGE,
- extra);
- } else {
- status_t err = mTSParser->feedTSPacket(buffer, sizeof(buffer));
-
- if (err != OK) {
- ALOGE("TS Parser returned error %d", err);
- mTSParser->signalEOS(err);
- mFinalResult = err;
- break;
- }
- }
-
- mOffset += n;
- }
- }
-
return OK;
}
status_t NuPlayer::HTTPLiveSource::dequeueAccessUnit(
bool audio, sp<ABuffer> *accessUnit) {
- ATSParser::SourceType type =
- audio ? ATSParser::AUDIO : ATSParser::VIDEO;
-
- sp<AnotherPacketSource> source =
- static_cast<AnotherPacketSource *>(mTSParser->getSource(type).get());
-
- if (source == NULL) {
- return -EWOULDBLOCK;
- }
-
- status_t finalResult;
- if (!source->hasBufferAvailable(&finalResult)) {
- return finalResult == OK ? -EWOULDBLOCK : finalResult;
- }
-
- return source->dequeueAccessUnit(accessUnit);
+ return mLiveSession->dequeueAccessUnit(
+ audio ? LiveSession::STREAMTYPE_AUDIO
+ : LiveSession::STREAMTYPE_VIDEO,
+ accessUnit);
}
status_t NuPlayer::HTTPLiveSource::getDuration(int64_t *durationUs) {
@@ -189,15 +121,7 @@
}
status_t NuPlayer::HTTPLiveSource::seekTo(int64_t seekTimeUs) {
- // We need to make sure we're not seeking until we have seen the very first
- // PTS timestamp in the whole stream (from the beginning of the stream).
- while (!mTSParser->PTSTimeDeltaEstablished() && feedMoreTSData() == OK) {
- usleep(100000);
- }
-
- mLiveSession->seekTo(seekTimeUs);
-
- return OK;
+ return mLiveSession->seekTo(seekTimeUs);
}
void NuPlayer::HTTPLiveSource::onMessageReceived(const sp<AMessage> &msg) {
@@ -249,6 +173,32 @@
break;
}
+ case LiveSession::kWhatStreamsChanged:
+ {
+ uint32_t changedMask;
+ CHECK(msg->findInt32(
+ "changedMask", (int32_t *)&changedMask));
+
+ bool audio = changedMask & LiveSession::STREAMTYPE_AUDIO;
+ bool video = changedMask & LiveSession::STREAMTYPE_VIDEO;
+
+ sp<AMessage> reply;
+ CHECK(msg->findMessage("reply", &reply));
+
+ sp<AMessage> notify = dupNotify();
+ notify->setInt32("what", kWhatQueueDecoderShutdown);
+ notify->setInt32("audio", audio);
+ notify->setInt32("video", video);
+ notify->setMessage("reply", reply);
+ notify->post();
+ break;
+ }
+
+ case LiveSession::kWhatError:
+ {
+ break;
+ }
+
default:
TRESPASS();
}
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
index 067d1da..aa9434b 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
@@ -23,7 +23,6 @@
namespace android {
-struct ATSParser;
struct LiveSession;
struct NuPlayer::HTTPLiveSource : public NuPlayer::Source {
@@ -37,18 +36,16 @@
virtual void prepareAsync();
virtual void start();
- virtual status_t feedMoreTSData();
-
virtual status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit);
+ virtual sp<AMessage> getFormat(bool audio);
+ virtual status_t feedMoreTSData();
virtual status_t getDuration(int64_t *durationUs);
virtual status_t seekTo(int64_t seekTimeUs);
protected:
virtual ~HTTPLiveSource();
- virtual sp<MetaData> getFormatMeta(bool audio);
-
virtual void onMessageReceived(const sp<AMessage> &msg);
private:
@@ -70,7 +67,6 @@
off64_t mOffset;
sp<ALooper> mLiveLooper;
sp<LiveSession> mLiveSession;
- sp<ATSParser> mTSParser;
void onSessionNotify(const sp<AMessage> &msg);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index b89b1c8..7e81035 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -89,6 +89,38 @@
DISALLOW_EVIL_CONSTRUCTORS(SetSurfaceAction);
};
+struct NuPlayer::ShutdownDecoderAction : public Action {
+ ShutdownDecoderAction(bool audio, bool video)
+ : mAudio(audio),
+ mVideo(video) {
+ }
+
+ virtual void execute(NuPlayer *player) {
+ player->performDecoderShutdown(mAudio, mVideo);
+ }
+
+private:
+ bool mAudio;
+ bool mVideo;
+
+ DISALLOW_EVIL_CONSTRUCTORS(ShutdownDecoderAction);
+};
+
+struct NuPlayer::PostMessageAction : public Action {
+ PostMessageAction(const sp<AMessage> &msg)
+ : mMessage(msg) {
+ }
+
+ virtual void execute(NuPlayer *) {
+ mMessage->post();
+ }
+
+private:
+ sp<AMessage> mMessage;
+
+ DISALLOW_EVIL_CONSTRUCTORS(PostMessageAction);
+};
+
// Use this if there's no state necessary to save in order to execute
// the action.
struct NuPlayer::SimpleAction : public Action {
@@ -335,7 +367,8 @@
ALOGV("kWhatSetVideoNativeWindow");
mDeferredActions.push_back(
- new SimpleAction(&NuPlayer::performDecoderShutdown));
+ new ShutdownDecoderAction(
+ false /* audio */, true /* video */));
sp<RefBase> obj;
CHECK(msg->findObject("native-window", &obj));
@@ -712,7 +745,8 @@
ALOGV("kWhatReset");
mDeferredActions.push_back(
- new SimpleAction(&NuPlayer::performDecoderShutdown));
+ new ShutdownDecoderAction(
+ true /* audio */, true /* video */));
mDeferredActions.push_back(
new SimpleAction(&NuPlayer::performReset));
@@ -1023,6 +1057,9 @@
}
void NuPlayer::flushDecoder(bool audio, bool needShutdown) {
+ ALOGV("[%s] flushDecoder needShutdown=%d",
+ audio ? "audio" : "video", needShutdown);
+
if ((audio && mAudioDecoder == NULL) || (!audio && mVideoDecoder == NULL)) {
ALOGI("flushDecoder %s without decoder present",
audio ? "audio" : "video");
@@ -1173,20 +1210,29 @@
}
}
-void NuPlayer::performDecoderShutdown() {
- ALOGV("performDecoderShutdown");
+void NuPlayer::performDecoderShutdown(bool audio, bool video) {
+ ALOGV("performDecoderShutdown audio=%d, video=%d", audio, video);
- if (mAudioDecoder == NULL && mVideoDecoder == NULL) {
+ if ((!audio || mAudioDecoder == NULL)
+ && (!video || mVideoDecoder == NULL)) {
return;
}
mTimeDiscontinuityPending = true;
- if (mAudioDecoder != NULL) {
+ if (mFlushingAudio == NONE && (!audio || mAudioDecoder == NULL)) {
+ mFlushingAudio = FLUSHED;
+ }
+
+ if (mFlushingVideo == NONE && (!video || mVideoDecoder == NULL)) {
+ mFlushingVideo = FLUSHED;
+ }
+
+ if (audio && mAudioDecoder != NULL) {
flushDecoder(true /* audio */, true /* needShutdown */);
}
- if (mVideoDecoder != NULL) {
+ if (video && mVideoDecoder != NULL) {
flushDecoder(false /* audio */, true /* needShutdown */);
}
}
@@ -1322,6 +1368,19 @@
break;
}
+ case Source::kWhatQueueDecoderShutdown:
+ {
+ int32_t audio, video;
+ CHECK(msg->findInt32("audio", &audio));
+ CHECK(msg->findInt32("video", &video));
+
+ sp<AMessage> reply;
+ CHECK(msg->findMessage("reply", &reply));
+
+ queueDecoderShutdown(audio, video, reply);
+ break;
+ }
+
default:
TRESPASS();
}
@@ -1355,4 +1414,19 @@
TRESPASS();
}
+void NuPlayer::queueDecoderShutdown(
+ bool audio, bool video, const sp<AMessage> &reply) {
+ ALOGI("queueDecoderShutdown audio=%d, video=%d", audio, video);
+
+ mDeferredActions.push_back(
+ new ShutdownDecoderAction(audio, video));
+
+ mDeferredActions.push_back(
+ new SimpleAction(&NuPlayer::performScanSources));
+
+ mDeferredActions.push_back(new PostMessageAction(reply));
+
+ processDeferredActions();
+}
+
} // namespace android
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 50d0462..8b6c8c1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -80,6 +80,8 @@
struct Action;
struct SeekAction;
struct SetSurfaceAction;
+ struct ShutdownDecoderAction;
+ struct PostMessageAction;
struct SimpleAction;
enum {
@@ -172,13 +174,16 @@
void performSeek(int64_t seekTimeUs);
void performDecoderFlush();
- void performDecoderShutdown();
+ void performDecoderShutdown(bool audio, bool video);
void performReset();
void performScanSources();
void performSetSurface(const sp<NativeWindowWrapper> &wrapper);
void onSourceNotify(const sp<AMessage> &msg);
+ void queueDecoderShutdown(
+ bool audio, bool video, const sp<AMessage> &reply);
+
DISALLOW_EVIL_CONSTRUCTORS(NuPlayer);
};
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 404b56f..b543d9d 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -95,11 +95,11 @@
}
void NuPlayer::Renderer::signalTimeDiscontinuity() {
- CHECK(mAudioQueue.empty());
- CHECK(mVideoQueue.empty());
+ // CHECK(mAudioQueue.empty());
+ // CHECK(mVideoQueue.empty());
mAnchorTimeMediaUs = -1;
mAnchorTimeRealUs = -1;
- mSyncQueues = mHasAudio && mHasVideo;
+ mSyncQueues = false;
}
void NuPlayer::Renderer::pause() {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
index 1cbf575..81ffd21 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
@@ -42,6 +42,7 @@
kWhatVideoSizeChanged,
kWhatBufferingStart,
kWhatBufferingEnd,
+ kWhatQueueDecoderShutdown,
};
// The provides message is used to notify the player about various
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index cf41cf2..bf650b4 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -359,6 +359,7 @@
mNode(NULL),
mSentFormat(false),
mIsEncoder(false),
+ mUseMetadataOnEncoderOutput(false),
mShutdownInProgress(false),
mEncoderDelay(0),
mEncoderPadding(0),
@@ -483,7 +484,8 @@
? OMXCodec::kRequiresAllocateBufferOnInputPorts
: OMXCodec::kRequiresAllocateBufferOnOutputPorts;
- if (portIndex == kPortIndexInput && (mFlags & kFlagIsSecure)) {
+ if ((portIndex == kPortIndexInput && (mFlags & kFlagIsSecure))
+ || mUseMetadataOnEncoderOutput) {
mem.clear();
void *ptr;
@@ -491,7 +493,10 @@
mNode, portIndex, def.nBufferSize, &info.mBufferID,
&ptr);
- info.mData = new ABuffer(ptr, def.nBufferSize);
+ int32_t bufSize = mUseMetadataOnEncoderOutput ?
+ (4 + sizeof(buffer_handle_t)) : def.nBufferSize;
+
+ info.mData = new ABuffer(ptr, bufSize);
} else if (mQuirks & requiresAllocateBufferBit) {
err = mOMX->allocateBufferWithBackup(
mNode, portIndex, mem, &info.mBufferID);
@@ -912,14 +917,14 @@
err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexInput, OMX_TRUE);
if (err != OK) {
- ALOGE("[%s] storeMetaDataInBuffers failed w/ err %d",
- mComponentName.c_str(), err);
+ ALOGE("[%s] storeMetaDataInBuffers (input) failed w/ err %d",
+ mComponentName.c_str(), err);
- return err;
- }
- }
+ return err;
+ }
+ }
- int32_t prependSPSPPS;
+ int32_t prependSPSPPS = 0;
if (encoder
&& msg->findInt32("prepend-sps-pps-to-idr-frames", &prependSPSPPS)
&& prependSPSPPS != 0) {
@@ -946,7 +951,27 @@
}
}
- if (!strncasecmp(mime, "video/", 6)) {
+ // Only enable metadata mode on encoder output if encoder can prepend
+ // sps/pps to idr frames, since in metadata mode the bitstream is in an
+ // opaque handle, to which we don't have access.
+ int32_t video = !strncasecmp(mime, "video/", 6);
+ if (encoder && video) {
+ OMX_BOOL enable = (OMX_BOOL) (prependSPSPPS
+ && msg->findInt32("store-metadata-in-buffers-output", &storeMeta)
+ && storeMeta != 0);
+
+ err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, enable);
+
+ if (err != OK) {
+ ALOGE("[%s] storeMetaDataInBuffers (output) failed w/ err %d",
+ mComponentName.c_str(), err);
+ mUseMetadataOnEncoderOutput = 0;
+ } else {
+ mUseMetadataOnEncoderOutput = enable;
+ }
+ }
+
+ if (video) {
if (encoder) {
err = setupVideoEncoder(mime, msg);
} else {
@@ -2321,10 +2346,15 @@
¶ms, sizeof(params)),
(status_t)OK);
+ CHECK_GT(params.nChannels, 0);
CHECK(params.nChannels == 1 || params.bInterleaved);
CHECK_EQ(params.nBitPerSample, 16u);
- CHECK_EQ((int)params.eNumData, (int)OMX_NumericalDataSigned);
- CHECK_EQ((int)params.ePCMMode, (int)OMX_AUDIO_PCMModeLinear);
+
+ CHECK_EQ((int)params.eNumData,
+ (int)OMX_NumericalDataSigned);
+
+ CHECK_EQ((int)params.ePCMMode,
+ (int)OMX_AUDIO_PCMModeLinear);
notify->setString("mime", MEDIA_MIMETYPE_AUDIO_RAW);
notify->setInt32("channel-count", params.nChannels);
@@ -2334,11 +2364,14 @@
if (mSkipCutBuffer != NULL) {
size_t prevbufsize = mSkipCutBuffer->size();
if (prevbufsize != 0) {
- ALOGW("Replacing SkipCutBuffer holding %d bytes", prevbufsize);
+ ALOGW("Replacing SkipCutBuffer holding %d "
+ "bytes",
+ prevbufsize);
}
}
- mSkipCutBuffer = new SkipCutBuffer(mEncoderDelay * frameSize,
- mEncoderPadding * frameSize);
+ mSkipCutBuffer = new SkipCutBuffer(
+ mEncoderDelay * frameSize,
+ mEncoderPadding * frameSize);
}
if (mChannelMaskPresent) {
@@ -3062,7 +3095,15 @@
mCodec->sendFormatChange(reply);
}
- info->mData->setRange(rangeOffset, rangeLength);
+ if (mCodec->mUseMetadataOnEncoderOutput) {
+ native_handle_t* handle =
+ *(native_handle_t**)(info->mData->data() + 4);
+ info->mData->meta()->setPointer("handle", handle);
+ info->mData->meta()->setInt32("rangeOffset", rangeOffset);
+ info->mData->meta()->setInt32("rangeLength", rangeLength);
+ } else {
+ info->mData->setRange(rangeOffset, rangeLength);
+ }
#if 0
if (mCodec->mNativeWindow == NULL) {
if (IsIDR(info->mData)) {
@@ -3220,6 +3261,7 @@
mCodec->mOMX.clear();
mCodec->mQuirks = 0;
mCodec->mFlags = 0;
+ mCodec->mUseMetadataOnEncoderOutput = 0;
mCodec->mComponentName.clear();
}
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index acc3abf..9544dbc 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -69,7 +69,6 @@
LOCAL_SHARED_LIBRARIES := \
libbinder \
libcamera_client \
- libcrypto \
libcutils \
libdl \
libdrmframework \
@@ -97,7 +96,6 @@
libvpx \
libwebm \
libstagefright_mpeg2ts \
- libstagefright_httplive \
libstagefright_id3 \
libFLAC \
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 4208019..92efae8 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -36,8 +36,7 @@
const sp<MediaPlayerBase::AudioSink> &audioSink,
bool allowDeepBuffering,
AwesomePlayer *observer)
- : mAudioTrack(NULL),
- mInputBuffer(NULL),
+ : mInputBuffer(NULL),
mSampleRate(0),
mLatencyUs(0),
mFrameSize(0),
@@ -166,8 +165,7 @@
0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);
if ((err = mAudioTrack->initCheck()) != OK) {
- delete mAudioTrack;
- mAudioTrack = NULL;
+ mAudioTrack.clear();
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
@@ -235,8 +233,7 @@
} else {
mAudioTrack->stop();
- delete mAudioTrack;
- mAudioTrack = NULL;
+ mAudioTrack.clear();
}
// Make sure to release any buffer we hold onto so that the
@@ -297,7 +294,7 @@
status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
if (mAudioSink.get() != NULL) {
return mAudioSink->setPlaybackRatePermille(ratePermille);
- } else if (mAudioTrack != NULL){
+ } else if (mAudioTrack != 0){
return mAudioTrack->setSampleRate(ratePermille * mSampleRate / 1000);
} else {
return NO_INIT;
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 3cf4d5c..bdd842f 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -49,8 +49,7 @@
AudioSource::AudioSource(
audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount)
- : mRecord(NULL),
- mStarted(false),
+ : mStarted(false),
mSampleRate(sampleRate),
mPrevSampleTimeUs(0),
mNumFramesReceived(0),
@@ -91,9 +90,6 @@
if (mStarted) {
reset();
}
-
- delete mRecord;
- mRecord = NULL;
}
status_t AudioSource::initCheck() const {
@@ -122,8 +118,7 @@
if (err == OK) {
mStarted = true;
} else {
- delete mRecord;
- mRecord = NULL;
+ mRecord.clear();
}
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index bd28118..6c197e2 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -597,7 +597,7 @@
bool AwesomePlayer::getBitrate(int64_t *bitrate) {
off64_t size;
- if (mDurationUs >= 0 && mCachedSource != NULL
+ if (mDurationUs > 0 && mCachedSource != NULL
&& mCachedSource->getSize(&size) == OK) {
*bitrate = size * 8000000ll / mDurationUs; // in bits/sec
return true;
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 145869e..42a9c7a 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -341,6 +341,7 @@
mDataSource(source),
mInitCheck(NO_INIT),
mHasVideo(false),
+ mHeaderTimescale(0),
mFirstTrack(NULL),
mLastTrack(NULL),
mFileMetaData(new MetaData),
@@ -817,6 +818,7 @@
case FOURCC('i', 'l', 's', 't'):
case FOURCC('s', 'i', 'n', 'f'):
case FOURCC('s', 'c', 'h', 'i'):
+ case FOURCC('e', 'd', 't', 's'):
{
if (chunk_type == FOURCC('s', 't', 'b', 'l')) {
ALOGV("sampleTable chunk is %d bytes long.", (size_t)chunk_size);
@@ -904,6 +906,68 @@
break;
}
+ case FOURCC('e', 'l', 's', 't'):
+ {
+ // See 14496-12 8.6.6
+ uint8_t version;
+ if (mDataSource->readAt(data_offset, &version, 1) < 1) {
+ return ERROR_IO;
+ }
+
+ uint32_t entry_count;
+ if (!mDataSource->getUInt32(data_offset + 4, &entry_count)) {
+ return ERROR_IO;
+ }
+
+ if (entry_count != 1) {
+ // we only support a single entry at the moment, for gapless playback
+ ALOGW("ignoring edit list with %d entries", entry_count);
+ } else if (mHeaderTimescale == 0) {
+ ALOGW("ignoring edit list because timescale is 0");
+ } else {
+ off64_t entriesoffset = data_offset + 8;
+ uint64_t segment_duration;
+ int64_t media_time;
+
+ if (version == 1) {
+ if (!mDataSource->getUInt64(entriesoffset, &segment_duration) ||
+ !mDataSource->getUInt64(entriesoffset + 8, (uint64_t*)&media_time)) {
+ return ERROR_IO;
+ }
+ } else if (version == 0) {
+ uint32_t sd;
+ int32_t mt;
+ if (!mDataSource->getUInt32(entriesoffset, &sd) ||
+ !mDataSource->getUInt32(entriesoffset + 4, (uint32_t*)&mt)) {
+ return ERROR_IO;
+ }
+ segment_duration = sd;
+ media_time = mt;
+ } else {
+ return ERROR_IO;
+ }
+
+ uint64_t halfscale = mHeaderTimescale / 2;
+ segment_duration = (segment_duration * 1000000 + halfscale)/ mHeaderTimescale;
+ media_time = (media_time * 1000000 + halfscale) / mHeaderTimescale;
+
+ int64_t duration;
+ int32_t samplerate;
+ if (mLastTrack->meta->findInt64(kKeyDuration, &duration) &&
+ mLastTrack->meta->findInt32(kKeySampleRate, &samplerate)) {
+
+ int64_t delay = (media_time * samplerate + 500000) / 1000000;
+ mLastTrack->meta->setInt32(kKeyEncoderDelay, delay);
+
+ int64_t paddingus = duration - (segment_duration + media_time);
+ int64_t paddingsamples = (paddingus * samplerate + 500000) / 1000000;
+ mLastTrack->meta->setInt32(kKeyEncoderPadding, paddingsamples);
+ }
+ }
+ *offset += chunk_size;
+ break;
+ }
+
case FOURCC('f', 'r', 'm', 'a'):
{
uint32_t original_fourcc;
@@ -1564,24 +1628,26 @@
case FOURCC('m', 'v', 'h', 'd'):
{
- if (chunk_data_size < 12) {
+ if (chunk_data_size < 24) {
return ERROR_MALFORMED;
}
- uint8_t header[12];
+ uint8_t header[24];
if (mDataSource->readAt(
data_offset, header, sizeof(header))
< (ssize_t)sizeof(header)) {
return ERROR_IO;
}
- int64_t creationTime;
+ uint64_t creationTime;
if (header[0] == 1) {
creationTime = U64_AT(&header[4]);
+ mHeaderTimescale = U32_AT(&header[20]);
} else if (header[0] != 0) {
return ERROR_MALFORMED;
} else {
creationTime = U32_AT(&header[4]);
+ mHeaderTimescale = U32_AT(&header[12]);
}
String8 s;
diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 409038a..71b6569 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -305,8 +305,9 @@
// First time seeing the buffer? Added it to the SMS slot
if (item.mGraphicBuffer != NULL) {
- mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+ mSlots[item.mBuf].mGraphicBuffer = item.mGraphicBuffer;
}
+ mSlots[item.mBuf].mFrameNumber = item.mFrameNumber;
// check for the timing of this buffer
if (mNumFramesReceived == 0 && !mUseAbsoluteTimestamps) {
@@ -315,7 +316,8 @@
if (mStartTimeNs > 0) {
if (item.mTimestamp < mStartTimeNs) {
// This frame predates start of record, discard
- mBufferQueue->releaseBuffer(item.mBuf, EGL_NO_DISPLAY,
+ mBufferQueue->releaseBuffer(
+ item.mBuf, item.mFrameNumber, EGL_NO_DISPLAY,
EGL_NO_SYNC_KHR, Fence::NO_FENCE);
continue;
}
@@ -345,17 +347,18 @@
// First time seeing the buffer? Added it to the SMS slot
if (item.mGraphicBuffer != NULL) {
- mBufferSlot[mCurrentSlot] = item.mGraphicBuffer;
+ mSlots[item.mBuf].mGraphicBuffer = item.mGraphicBuffer;
}
+ mSlots[item.mBuf].mFrameNumber = item.mFrameNumber;
- mCurrentBuffers.push_back(mBufferSlot[mCurrentSlot]);
+ mCurrentBuffers.push_back(mSlots[mCurrentSlot].mGraphicBuffer);
int64_t prevTimeStamp = mCurrentTimestamp;
mCurrentTimestamp = item.mTimestamp;
mNumFramesEncoded++;
// Pass the data to the MediaBuffer. Pass in only the metadata
- passMetadataBuffer(buffer, mBufferSlot[mCurrentSlot]->handle);
+ passMetadataBuffer(buffer, mSlots[mCurrentSlot].mGraphicBuffer->handle);
(*buffer)->setObserver(this);
(*buffer)->add_ref();
@@ -405,15 +408,16 @@
}
for (int id = 0; id < BufferQueue::NUM_BUFFER_SLOTS; id++) {
- if (mBufferSlot[id] == NULL) {
+ if (mSlots[id].mGraphicBuffer == NULL) {
continue;
}
- if (bufferHandle == mBufferSlot[id]->handle) {
+ if (bufferHandle == mSlots[id].mGraphicBuffer->handle) {
ALOGV("Slot %d returned, matches handle = %p", id,
- mBufferSlot[id]->handle);
+ mSlots[id].mGraphicBuffer->handle);
- mBufferQueue->releaseBuffer(id, EGL_NO_DISPLAY, EGL_NO_SYNC_KHR,
+ mBufferQueue->releaseBuffer(id, mSlots[id].mFrameNumber,
+ EGL_NO_DISPLAY, EGL_NO_SYNC_KHR,
Fence::NO_FENCE);
buffer->setObserver(0);
@@ -469,7 +473,7 @@
mFrameAvailableCondition.signal();
for (int i = 0; i < BufferQueue::NUM_BUFFER_SLOTS; i++) {
- mBufferSlot[i] = 0;
+ mSlots[i].mGraphicBuffer = 0;
}
}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index cf50dc9..1b20cbb 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -604,6 +604,9 @@
// To make the codec behave the same before and after a reset, we need to invalidate the
// streaminfo struct. This does that:
mStreamInfo->sampleRate = 0;
+
+ mSignalledError = false;
+ mOutputPortSettingsChange = NONE;
}
void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
diff --git a/media/libstagefright/codecs/aacenc/SampleCode/Android.mk b/media/libstagefright/codecs/aacenc/SampleCode/Android.mk
index 01016e7..d06dcf6 100644
--- a/media/libstagefright/codecs/aacenc/SampleCode/Android.mk
+++ b/media/libstagefright/codecs/aacenc/SampleCode/Android.mk
@@ -5,7 +5,7 @@
AAC_E_SAMPLES.c \
../../common/cmnMemory.c
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE := AACEncTest
diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
index 4d4212f..3320688 100644
--- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
+++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
@@ -457,6 +457,11 @@
}
}
+void SoftAMR::onReset() {
+ mSignalledError = false;
+ mOutputPortSettingsChange = NONE;
+}
+
} // namespace android
android::SoftOMXComponent *createSoftOMXComponent(
diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.h b/media/libstagefright/codecs/amrnb/dec/SoftAMR.h
index 9a596e5..758d6ac 100644
--- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.h
+++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.h
@@ -40,6 +40,7 @@
virtual void onQueueFilled(OMX_U32 portIndex);
virtual void onPortFlushCompleted(OMX_U32 portIndex);
virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled);
+ virtual void onReset();
private:
enum {
diff --git a/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk b/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk
index db34d08..c203f77 100644
--- a/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk
+++ b/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk
@@ -5,7 +5,7 @@
AMRWB_E_SAMPLE.c \
../../common/cmnMemory.c
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE := AMRWBEncTest
LOCAL_ARM_MODE := arm
diff --git a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
index 020cc0a..fb2a430 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
@@ -48,42 +48,32 @@
{ OMX_VIDEO_H263ProfileISWV2, OMX_VIDEO_H263Level45 },
};
-template<class T>
-static void InitOMXParams(T *params) {
- params->nSize = sizeof(T);
- params->nVersion.s.nVersionMajor = 1;
- params->nVersion.s.nVersionMinor = 0;
- params->nVersion.s.nRevision = 0;
- params->nVersion.s.nStep = 0;
-}
-
SoftMPEG4::SoftMPEG4(
const char *name,
+ const char *componentRole,
+ OMX_VIDEO_CODINGTYPE codingType,
+ const CodecProfileLevel *profileLevels,
+ size_t numProfileLevels,
const OMX_CALLBACKTYPE *callbacks,
OMX_PTR appData,
OMX_COMPONENTTYPE **component)
- : SimpleSoftOMXComponent(name, callbacks, appData, component),
- mMode(MODE_MPEG4),
+ : SoftVideoDecoderOMXComponent(
+ name, componentRole, codingType, profileLevels, numProfileLevels,
+ 352 /* width */, 288 /* height */, callbacks, appData, component),
+ mMode(codingType == OMX_VIDEO_CodingH263 ? MODE_H263 : MODE_MPEG4),
mHandle(new tagvideoDecControls),
mInputBufferCount(0),
- mWidth(352),
- mHeight(288),
- mCropLeft(0),
- mCropTop(0),
- mCropRight(mWidth - 1),
- mCropBottom(mHeight - 1),
mSignalledError(false),
mInitialized(false),
mFramesConfigured(false),
mNumSamplesOutput(0),
- mOutputPortSettingsChange(NONE) {
- if (!strcmp(name, "OMX.google.h263.decoder")) {
- mMode = MODE_H263;
- } else {
- CHECK(!strcmp(name, "OMX.google.mpeg4.decoder"));
- }
-
- initPorts();
+ mPvTime(0) {
+ initPorts(
+ kNumInputBuffers,
+ 8192 /* inputBufferSize */,
+ kNumOutputBuffers,
+ (mMode == MODE_MPEG4)
+ ? MEDIA_MIMETYPE_VIDEO_MPEG4 : MEDIA_MIMETYPE_VIDEO_H263);
CHECK_EQ(initDecoder(), (status_t)OK);
}
@@ -96,219 +86,11 @@
mHandle = NULL;
}
-void SoftMPEG4::initPorts() {
- OMX_PARAM_PORTDEFINITIONTYPE def;
- InitOMXParams(&def);
-
- def.nPortIndex = 0;
- def.eDir = OMX_DirInput;
- def.nBufferCountMin = kNumInputBuffers;
- def.nBufferCountActual = def.nBufferCountMin;
- def.nBufferSize = 8192;
- def.bEnabled = OMX_TRUE;
- def.bPopulated = OMX_FALSE;
- def.eDomain = OMX_PortDomainVideo;
- def.bBuffersContiguous = OMX_FALSE;
- def.nBufferAlignment = 1;
-
- def.format.video.cMIMEType =
- (mMode == MODE_MPEG4)
- ? const_cast<char *>(MEDIA_MIMETYPE_VIDEO_MPEG4)
- : const_cast<char *>(MEDIA_MIMETYPE_VIDEO_H263);
-
- def.format.video.pNativeRender = NULL;
- def.format.video.nFrameWidth = mWidth;
- def.format.video.nFrameHeight = mHeight;
- def.format.video.nStride = def.format.video.nFrameWidth;
- def.format.video.nSliceHeight = def.format.video.nFrameHeight;
- def.format.video.nBitrate = 0;
- def.format.video.xFramerate = 0;
- def.format.video.bFlagErrorConcealment = OMX_FALSE;
-
- def.format.video.eCompressionFormat =
- mMode == MODE_MPEG4 ? OMX_VIDEO_CodingMPEG4 : OMX_VIDEO_CodingH263;
-
- def.format.video.eColorFormat = OMX_COLOR_FormatUnused;
- def.format.video.pNativeWindow = NULL;
-
- addPort(def);
-
- def.nPortIndex = 1;
- def.eDir = OMX_DirOutput;
- def.nBufferCountMin = kNumOutputBuffers;
- def.nBufferCountActual = def.nBufferCountMin;
- def.bEnabled = OMX_TRUE;
- def.bPopulated = OMX_FALSE;
- def.eDomain = OMX_PortDomainVideo;
- def.bBuffersContiguous = OMX_FALSE;
- def.nBufferAlignment = 2;
-
- def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_RAW);
- def.format.video.pNativeRender = NULL;
- def.format.video.nFrameWidth = mWidth;
- def.format.video.nFrameHeight = mHeight;
- def.format.video.nStride = def.format.video.nFrameWidth;
- def.format.video.nSliceHeight = def.format.video.nFrameHeight;
- def.format.video.nBitrate = 0;
- def.format.video.xFramerate = 0;
- def.format.video.bFlagErrorConcealment = OMX_FALSE;
- def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused;
- def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar;
- def.format.video.pNativeWindow = NULL;
-
- def.nBufferSize =
- (def.format.video.nFrameWidth * def.format.video.nFrameHeight * 3) / 2;
-
- addPort(def);
-}
-
status_t SoftMPEG4::initDecoder() {
memset(mHandle, 0, sizeof(tagvideoDecControls));
return OK;
}
-OMX_ERRORTYPE SoftMPEG4::internalGetParameter(
- OMX_INDEXTYPE index, OMX_PTR params) {
- switch (index) {
- case OMX_IndexParamVideoPortFormat:
- {
- OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
- (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
-
- if (formatParams->nPortIndex > 1) {
- return OMX_ErrorUndefined;
- }
-
- if (formatParams->nIndex != 0) {
- return OMX_ErrorNoMore;
- }
-
- if (formatParams->nPortIndex == 0) {
- formatParams->eCompressionFormat =
- (mMode == MODE_MPEG4)
- ? OMX_VIDEO_CodingMPEG4 : OMX_VIDEO_CodingH263;
-
- formatParams->eColorFormat = OMX_COLOR_FormatUnused;
- formatParams->xFramerate = 0;
- } else {
- CHECK_EQ(formatParams->nPortIndex, 1u);
-
- formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused;
- formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar;
- formatParams->xFramerate = 0;
- }
-
- return OMX_ErrorNone;
- }
-
- case OMX_IndexParamVideoProfileLevelQuerySupported:
- {
- OMX_VIDEO_PARAM_PROFILELEVELTYPE *profileLevel =
- (OMX_VIDEO_PARAM_PROFILELEVELTYPE *) params;
-
- if (profileLevel->nPortIndex != 0) { // Input port only
- ALOGE("Invalid port index: %ld", profileLevel->nPortIndex);
- return OMX_ErrorUnsupportedIndex;
- }
-
- size_t index = profileLevel->nProfileIndex;
- if (mMode == MODE_H263) {
- size_t nProfileLevels =
- sizeof(kH263ProfileLevels) / sizeof(kH263ProfileLevels[0]);
- if (index >= nProfileLevels) {
- return OMX_ErrorNoMore;
- }
-
- profileLevel->eProfile = kH263ProfileLevels[index].mProfile;
- profileLevel->eLevel = kH263ProfileLevels[index].mLevel;
- } else {
- size_t nProfileLevels =
- sizeof(kM4VProfileLevels) / sizeof(kM4VProfileLevels[0]);
- if (index >= nProfileLevels) {
- return OMX_ErrorNoMore;
- }
-
- profileLevel->eProfile = kM4VProfileLevels[index].mProfile;
- profileLevel->eLevel = kM4VProfileLevels[index].mLevel;
- }
- return OMX_ErrorNone;
- }
-
- default:
- return SimpleSoftOMXComponent::internalGetParameter(index, params);
- }
-}
-
-OMX_ERRORTYPE SoftMPEG4::internalSetParameter(
- OMX_INDEXTYPE index, const OMX_PTR params) {
- switch (index) {
- case OMX_IndexParamStandardComponentRole:
- {
- const OMX_PARAM_COMPONENTROLETYPE *roleParams =
- (const OMX_PARAM_COMPONENTROLETYPE *)params;
-
- if (mMode == MODE_MPEG4) {
- if (strncmp((const char *)roleParams->cRole,
- "video_decoder.mpeg4",
- OMX_MAX_STRINGNAME_SIZE - 1)) {
- return OMX_ErrorUndefined;
- }
- } else {
- if (strncmp((const char *)roleParams->cRole,
- "video_decoder.h263",
- OMX_MAX_STRINGNAME_SIZE - 1)) {
- return OMX_ErrorUndefined;
- }
- }
-
- return OMX_ErrorNone;
- }
-
- case OMX_IndexParamVideoPortFormat:
- {
- OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
- (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
-
- if (formatParams->nPortIndex > 1) {
- return OMX_ErrorUndefined;
- }
-
- if (formatParams->nIndex != 0) {
- return OMX_ErrorNoMore;
- }
-
- return OMX_ErrorNone;
- }
-
- default:
- return SimpleSoftOMXComponent::internalSetParameter(index, params);
- }
-}
-
-OMX_ERRORTYPE SoftMPEG4::getConfig(
- OMX_INDEXTYPE index, OMX_PTR params) {
- switch (index) {
- case OMX_IndexConfigCommonOutputCrop:
- {
- OMX_CONFIG_RECTTYPE *rectParams = (OMX_CONFIG_RECTTYPE *)params;
-
- if (rectParams->nPortIndex != 1) {
- return OMX_ErrorUndefined;
- }
-
- rectParams->nLeft = mCropLeft;
- rectParams->nTop = mCropTop;
- rectParams->nWidth = mCropRight - mCropLeft + 1;
- rectParams->nHeight = mCropBottom - mCropTop + 1;
-
- return OMX_ErrorNone;
- }
-
- default:
- return OMX_ErrorUnsupportedIndex;
- }
-}
-
void SoftMPEG4::onQueueFilled(OMX_U32 portIndex) {
if (mSignalledError || mOutputPortSettingsChange != NONE) {
return;
@@ -415,9 +197,14 @@
uint32_t useExtTimestamp = (inHeader->nOffset == 0);
- // decoder deals in ms, OMX in us.
- uint32_t timestamp =
- useExtTimestamp ? (inHeader->nTimeStamp + 500) / 1000 : 0xFFFFFFFF;
+ // decoder deals in ms (int32_t), OMX in us (int64_t)
+ // so use fake timestamp instead
+ uint32_t timestamp = 0xFFFFFFFF;
+ if (useExtTimestamp) {
+ mPvToOmxTimeMap.add(mPvTime, inHeader->nTimeStamp);
+ timestamp = mPvTime;
+ mPvTime++;
+ }
int32_t bufferSize = inHeader->nFilledLen;
int32_t tmp = bufferSize;
@@ -441,7 +228,8 @@
}
// decoder deals in ms, OMX in us.
- outHeader->nTimeStamp = timestamp * 1000;
+ outHeader->nTimeStamp = mPvToOmxTimeMap.valueFor(timestamp);
+ mPvToOmxTimeMap.removeItem(timestamp);
inHeader->nOffset += bufferSize;
inHeader->nFilledLen = 0;
@@ -482,11 +270,11 @@
}
bool SoftMPEG4::portSettingsChanged() {
- int32_t disp_width, disp_height;
- PVGetVideoDimensions(mHandle, &disp_width, &disp_height);
+ uint32_t disp_width, disp_height;
+ PVGetVideoDimensions(mHandle, (int32 *)&disp_width, (int32 *)&disp_height);
- int32_t buf_width, buf_height;
- PVGetBufferDimensions(mHandle, &buf_width, &buf_height);
+ uint32_t buf_width, buf_height;
+ PVGetBufferDimensions(mHandle, (int32 *)&buf_width, (int32 *)&buf_height);
CHECK_LE(disp_width, buf_width);
CHECK_LE(disp_height, buf_height);
@@ -494,12 +282,12 @@
ALOGV("disp_width = %d, disp_height = %d, buf_width = %d, buf_height = %d",
disp_width, disp_height, buf_width, buf_height);
- if (mCropRight != disp_width - 1
- || mCropBottom != disp_height - 1) {
+ if (mCropWidth != disp_width
+ || mCropHeight != disp_height) {
mCropLeft = 0;
mCropTop = 0;
- mCropRight = disp_width - 1;
- mCropBottom = disp_height - 1;
+ mCropWidth = disp_width;
+ mCropHeight = disp_height;
notify(OMX_EventPortSettingsChanged,
1,
@@ -545,45 +333,22 @@
}
}
-void SoftMPEG4::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
- if (portIndex != 1) {
- return;
- }
-
- switch (mOutputPortSettingsChange) {
- case NONE:
- break;
-
- case AWAITING_DISABLED:
- {
- CHECK(!enabled);
- mOutputPortSettingsChange = AWAITING_ENABLED;
- break;
- }
-
- default:
- {
- CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
- CHECK(enabled);
- mOutputPortSettingsChange = NONE;
- break;
- }
+void SoftMPEG4::onReset() {
+ SoftVideoDecoderOMXComponent::onReset();
+ mPvToOmxTimeMap.clear();
+ mSignalledError = false;
+ mFramesConfigured = false;
+ if (mInitialized) {
+ PVCleanUpVideoDecoder(mHandle);
+ mInitialized = false;
}
}
void SoftMPEG4::updatePortDefinitions() {
- OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(0)->mDef;
- def->format.video.nFrameWidth = mWidth;
- def->format.video.nFrameHeight = mHeight;
- def->format.video.nStride = def->format.video.nFrameWidth;
- def->format.video.nSliceHeight = def->format.video.nFrameHeight;
+ SoftVideoDecoderOMXComponent::updatePortDefinitions();
- def = &editPortInfo(1)->mDef;
- def->format.video.nFrameWidth = mWidth;
- def->format.video.nFrameHeight = mHeight;
- def->format.video.nStride = def->format.video.nFrameWidth;
- def->format.video.nSliceHeight = def->format.video.nFrameHeight;
-
+ /* We have to align our width and height - this should affect stride! */
+ OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(kOutputPortIndex)->mDef;
def->nBufferSize =
(((def->format.video.nFrameWidth + 15) & -16)
* ((def->format.video.nFrameHeight + 15) & -16) * 3) / 2;
@@ -594,6 +359,19 @@
android::SoftOMXComponent *createSoftOMXComponent(
const char *name, const OMX_CALLBACKTYPE *callbacks,
OMX_PTR appData, OMX_COMPONENTTYPE **component) {
- return new android::SoftMPEG4(name, callbacks, appData, component);
+ using namespace android;
+ if (!strcmp(name, "OMX.google.h263.decoder")) {
+ return new android::SoftMPEG4(
+ name, "video_decoder.h263", OMX_VIDEO_CodingH263,
+ kH263ProfileLevels, ARRAY_SIZE(kH263ProfileLevels),
+ callbacks, appData, component);
+ } else if (!strcmp(name, "OMX.google.mpeg4.decoder")) {
+ return new android::SoftMPEG4(
+ name, "video_decoder.mpeg4", OMX_VIDEO_CodingMPEG4,
+ kM4VProfileLevels, ARRAY_SIZE(kM4VProfileLevels),
+ callbacks, appData, component);
+ } else {
+ CHECK(!"Unknown component");
+ }
}
diff --git a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h
index dff08a7..de14aaf 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h
+++ b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h
@@ -18,14 +18,18 @@
#define SOFT_MPEG4_H_
-#include "SimpleSoftOMXComponent.h"
+#include "SoftVideoDecoderOMXComponent.h"
struct tagvideoDecControls;
namespace android {
-struct SoftMPEG4 : public SimpleSoftOMXComponent {
+struct SoftMPEG4 : public SoftVideoDecoderOMXComponent {
SoftMPEG4(const char *name,
+ const char *componentRole,
+ OMX_VIDEO_CODINGTYPE codingType,
+ const CodecProfileLevel *profileLevels,
+ size_t numProfileLevels,
const OMX_CALLBACKTYPE *callbacks,
OMX_PTR appData,
OMX_COMPONENTTYPE **component);
@@ -33,17 +37,9 @@
protected:
virtual ~SoftMPEG4();
- virtual OMX_ERRORTYPE internalGetParameter(
- OMX_INDEXTYPE index, OMX_PTR params);
-
- virtual OMX_ERRORTYPE internalSetParameter(
- OMX_INDEXTYPE index, const OMX_PTR params);
-
- virtual OMX_ERRORTYPE getConfig(OMX_INDEXTYPE index, OMX_PTR params);
-
virtual void onQueueFilled(OMX_U32 portIndex);
virtual void onPortFlushCompleted(OMX_U32 portIndex);
- virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled);
+ virtual void onReset();
private:
enum {
@@ -54,32 +50,23 @@
enum {
MODE_MPEG4,
MODE_H263,
-
} mMode;
tagvideoDecControls *mHandle;
size_t mInputBufferCount;
- int32_t mWidth, mHeight;
- int32_t mCropLeft, mCropTop, mCropRight, mCropBottom;
-
bool mSignalledError;
bool mInitialized;
bool mFramesConfigured;
int32_t mNumSamplesOutput;
+ int32_t mPvTime;
+ KeyedVector<int32_t, OMX_TICKS> mPvToOmxTimeMap;
- enum {
- NONE,
- AWAITING_DISABLED,
- AWAITING_ENABLED
- } mOutputPortSettingsChange;
-
- void initPorts();
status_t initDecoder();
- void updatePortDefinitions();
+ virtual void updatePortDefinitions();
bool portSettingsChanged();
DISALLOW_EVIL_CONSTRUCTORS(SoftMPEG4);
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index 9f25536..7c382fb 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -361,6 +361,8 @@
void SoftMP3::onReset() {
pvmp3_InitDecoder(mConfig, mDecoderBuf);
mIsFirst = true;
+ mSignalledError = false;
+ mOutputPortSettingsChange = NONE;
}
} // namespace android
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index a400b4c..43d0263 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -29,26 +29,19 @@
namespace android {
-template<class T>
-static void InitOMXParams(T *params) {
- params->nSize = sizeof(T);
- params->nVersion.s.nVersionMajor = 1;
- params->nVersion.s.nVersionMinor = 0;
- params->nVersion.s.nRevision = 0;
- params->nVersion.s.nStep = 0;
-}
-
SoftVPX::SoftVPX(
const char *name,
const OMX_CALLBACKTYPE *callbacks,
OMX_PTR appData,
OMX_COMPONENTTYPE **component)
- : SimpleSoftOMXComponent(name, callbacks, appData, component),
- mCtx(NULL),
- mWidth(320),
- mHeight(240),
- mOutputPortSettingsChange(NONE) {
- initPorts();
+ : SoftVideoDecoderOMXComponent(
+ name, "video_decoder.vpx", OMX_VIDEO_CodingVPX,
+ NULL /* profileLevels */, 0 /* numProfileLevels */,
+ 320 /* width */, 240 /* height */, callbacks, appData, component),
+ mCtx(NULL) {
+ initPorts(kNumBuffers, 768 * 1024 /* inputBufferSize */,
+ kNumBuffers, MEDIA_MIMETYPE_VIDEO_VPX);
+
CHECK_EQ(initDecoder(), (status_t)OK);
}
@@ -58,65 +51,6 @@
mCtx = NULL;
}
-void SoftVPX::initPorts() {
- OMX_PARAM_PORTDEFINITIONTYPE def;
- InitOMXParams(&def);
-
- def.nPortIndex = 0;
- def.eDir = OMX_DirInput;
- def.nBufferCountMin = kNumBuffers;
- def.nBufferCountActual = def.nBufferCountMin;
- def.nBufferSize = 768 * 1024;
- def.bEnabled = OMX_TRUE;
- def.bPopulated = OMX_FALSE;
- def.eDomain = OMX_PortDomainVideo;
- def.bBuffersContiguous = OMX_FALSE;
- def.nBufferAlignment = 1;
-
- def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_VPX);
- def.format.video.pNativeRender = NULL;
- def.format.video.nFrameWidth = mWidth;
- def.format.video.nFrameHeight = mHeight;
- def.format.video.nStride = def.format.video.nFrameWidth;
- def.format.video.nSliceHeight = def.format.video.nFrameHeight;
- def.format.video.nBitrate = 0;
- def.format.video.xFramerate = 0;
- def.format.video.bFlagErrorConcealment = OMX_FALSE;
- def.format.video.eCompressionFormat = OMX_VIDEO_CodingVPX;
- def.format.video.eColorFormat = OMX_COLOR_FormatUnused;
- def.format.video.pNativeWindow = NULL;
-
- addPort(def);
-
- def.nPortIndex = 1;
- def.eDir = OMX_DirOutput;
- def.nBufferCountMin = kNumBuffers;
- def.nBufferCountActual = def.nBufferCountMin;
- def.bEnabled = OMX_TRUE;
- def.bPopulated = OMX_FALSE;
- def.eDomain = OMX_PortDomainVideo;
- def.bBuffersContiguous = OMX_FALSE;
- def.nBufferAlignment = 2;
-
- def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_RAW);
- def.format.video.pNativeRender = NULL;
- def.format.video.nFrameWidth = mWidth;
- def.format.video.nFrameHeight = mHeight;
- def.format.video.nStride = def.format.video.nFrameWidth;
- def.format.video.nSliceHeight = def.format.video.nFrameHeight;
- def.format.video.nBitrate = 0;
- def.format.video.xFramerate = 0;
- def.format.video.bFlagErrorConcealment = OMX_FALSE;
- def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused;
- def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar;
- def.format.video.pNativeWindow = NULL;
-
- def.nBufferSize =
- (def.format.video.nFrameWidth * def.format.video.nFrameHeight * 3) / 2;
-
- addPort(def);
-}
-
static int GetCPUCoreCount() {
int cpuCoreCount = 1;
#if defined(_SC_NPROCESSORS_ONLN)
@@ -145,80 +79,6 @@
return OK;
}
-OMX_ERRORTYPE SoftVPX::internalGetParameter(
- OMX_INDEXTYPE index, OMX_PTR params) {
- switch (index) {
- case OMX_IndexParamVideoPortFormat:
- {
- OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
- (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
-
- if (formatParams->nPortIndex > 1) {
- return OMX_ErrorUndefined;
- }
-
- if (formatParams->nIndex != 0) {
- return OMX_ErrorNoMore;
- }
-
- if (formatParams->nPortIndex == 0) {
- formatParams->eCompressionFormat = OMX_VIDEO_CodingVPX;
- formatParams->eColorFormat = OMX_COLOR_FormatUnused;
- formatParams->xFramerate = 0;
- } else {
- CHECK_EQ(formatParams->nPortIndex, 1u);
-
- formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused;
- formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar;
- formatParams->xFramerate = 0;
- }
-
- return OMX_ErrorNone;
- }
-
- default:
- return SimpleSoftOMXComponent::internalGetParameter(index, params);
- }
-}
-
-OMX_ERRORTYPE SoftVPX::internalSetParameter(
- OMX_INDEXTYPE index, const OMX_PTR params) {
- switch (index) {
- case OMX_IndexParamStandardComponentRole:
- {
- const OMX_PARAM_COMPONENTROLETYPE *roleParams =
- (const OMX_PARAM_COMPONENTROLETYPE *)params;
-
- if (strncmp((const char *)roleParams->cRole,
- "video_decoder.vpx",
- OMX_MAX_STRINGNAME_SIZE - 1)) {
- return OMX_ErrorUndefined;
- }
-
- return OMX_ErrorNone;
- }
-
- case OMX_IndexParamVideoPortFormat:
- {
- OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
- (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
-
- if (formatParams->nPortIndex > 1) {
- return OMX_ErrorUndefined;
- }
-
- if (formatParams->nIndex != 0) {
- return OMX_ErrorNoMore;
- }
-
- return OMX_ErrorNone;
- }
-
- default:
- return SimpleSoftOMXComponent::internalSetParameter(index, params);
- }
-}
-
void SoftVPX::onQueueFilled(OMX_U32 portIndex) {
if (mOutputPortSettingsChange != NONE) {
return;
@@ -226,6 +86,7 @@
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
+ bool EOSseen = false;
while (!inQueue.empty() && !outQueue.empty()) {
BufferInfo *inInfo = *inQueue.begin();
@@ -235,17 +96,20 @@
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ EOSseen = true;
+ if (inHeader->nFilledLen == 0) {
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
- outHeader->nFilledLen = 0;
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ outHeader->nFilledLen = 0;
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+ return;
+ }
}
if (vpx_codec_decode(
@@ -266,8 +130,8 @@
if (img != NULL) {
CHECK_EQ(img->fmt, IMG_FMT_I420);
- int32_t width = img->d_w;
- int32_t height = img->d_h;
+ uint32_t width = img->d_w;
+ uint32_t height = img->d_h;
if (width != mWidth || height != mHeight) {
mWidth = width;
@@ -282,7 +146,7 @@
outHeader->nOffset = 0;
outHeader->nFilledLen = (width * height * 3) / 2;
- outHeader->nFlags = 0;
+ outHeader->nFlags = EOSseen ? OMX_BUFFERFLAG_EOS : 0;
outHeader->nTimeStamp = inHeader->nTimeStamp;
const uint8_t *srcLine = (const uint8_t *)img->planes[PLANE_Y];
@@ -325,53 +189,6 @@
}
}
-void SoftVPX::onPortFlushCompleted(OMX_U32 portIndex) {
-}
-
-void SoftVPX::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
- if (portIndex != 1) {
- return;
- }
-
- switch (mOutputPortSettingsChange) {
- case NONE:
- break;
-
- case AWAITING_DISABLED:
- {
- CHECK(!enabled);
- mOutputPortSettingsChange = AWAITING_ENABLED;
- break;
- }
-
- default:
- {
- CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
- CHECK(enabled);
- mOutputPortSettingsChange = NONE;
- break;
- }
- }
-}
-
-void SoftVPX::updatePortDefinitions() {
- OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(0)->mDef;
- def->format.video.nFrameWidth = mWidth;
- def->format.video.nFrameHeight = mHeight;
- def->format.video.nStride = def->format.video.nFrameWidth;
- def->format.video.nSliceHeight = def->format.video.nFrameHeight;
-
- def = &editPortInfo(1)->mDef;
- def->format.video.nFrameWidth = mWidth;
- def->format.video.nFrameHeight = mHeight;
- def->format.video.nStride = def->format.video.nFrameWidth;
- def->format.video.nSliceHeight = def->format.video.nFrameHeight;
-
- def->nBufferSize =
- (def->format.video.nFrameWidth
- * def->format.video.nFrameHeight * 3) / 2;
-}
-
} // namespace android
android::SoftOMXComponent *createSoftOMXComponent(
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.h b/media/libstagefright/codecs/on2/dec/SoftVPX.h
index 3e814a2..626307b 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.h
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.h
@@ -18,11 +18,11 @@
#define SOFT_VPX_H_
-#include "SimpleSoftOMXComponent.h"
+#include "SoftVideoDecoderOMXComponent.h"
namespace android {
-struct SoftVPX : public SimpleSoftOMXComponent {
+struct SoftVPX : public SoftVideoDecoderOMXComponent {
SoftVPX(const char *name,
const OMX_CALLBACKTYPE *callbacks,
OMX_PTR appData,
@@ -31,15 +31,7 @@
protected:
virtual ~SoftVPX();
- virtual OMX_ERRORTYPE internalGetParameter(
- OMX_INDEXTYPE index, OMX_PTR params);
-
- virtual OMX_ERRORTYPE internalSetParameter(
- OMX_INDEXTYPE index, const OMX_PTR params);
-
virtual void onQueueFilled(OMX_U32 portIndex);
- virtual void onPortFlushCompleted(OMX_U32 portIndex);
- virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled);
private:
enum {
@@ -48,20 +40,8 @@
void *mCtx;
- int32_t mWidth;
- int32_t mHeight;
-
- enum {
- NONE,
- AWAITING_DISABLED,
- AWAITING_ENABLED
- } mOutputPortSettingsChange;
-
- void initPorts();
status_t initDecoder();
- void updatePortDefinitions();
-
DISALLOW_EVIL_CONSTRUCTORS(SoftVPX);
};
diff --git a/media/libstagefright/codecs/on2/h264dec/Android.mk b/media/libstagefright/codecs/on2/h264dec/Android.mk
index 2539f98..655b2ab 100644
--- a/media/libstagefright/codecs/on2/h264dec/Android.mk
+++ b/media/libstagefright/codecs/on2/h264dec/Android.mk
@@ -119,7 +119,7 @@
LOCAL_SHARED_LIBRARIES := libstagefright_soft_h264dec
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE := decoder
diff --git a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
index 6e36651..7ddb13c 100644
--- a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
+++ b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp
@@ -47,38 +47,28 @@
{ OMX_VIDEO_AVCProfileBaseline, OMX_VIDEO_AVCLevel51 },
};
-template<class T>
-static void InitOMXParams(T *params) {
- params->nSize = sizeof(T);
- params->nVersion.s.nVersionMajor = 1;
- params->nVersion.s.nVersionMinor = 0;
- params->nVersion.s.nRevision = 0;
- params->nVersion.s.nStep = 0;
-}
-
SoftAVC::SoftAVC(
const char *name,
const OMX_CALLBACKTYPE *callbacks,
OMX_PTR appData,
OMX_COMPONENTTYPE **component)
- : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ : SoftVideoDecoderOMXComponent(
+ name, "video_decoder.avc", OMX_VIDEO_CodingAVC,
+ kProfileLevels, ARRAY_SIZE(kProfileLevels),
+ 320 /* width */, 240 /* height */, callbacks, appData, component),
mHandle(NULL),
mInputBufferCount(0),
- mWidth(320),
- mHeight(240),
mPictureSize(mWidth * mHeight * 3 / 2),
- mCropLeft(0),
- mCropTop(0),
- mCropWidth(mWidth),
- mCropHeight(mHeight),
mFirstPicture(NULL),
mFirstPictureId(-1),
mPicId(0),
mHeadersDecoded(false),
mEOSStatus(INPUT_DATA_AVAILABLE),
- mOutputPortSettingsChange(NONE),
mSignalledError(false) {
- initPorts();
+ initPorts(
+ kNumInputBuffers, 8192 /* inputBufferSize */,
+ kNumOutputBuffers, MEDIA_MIMETYPE_VIDEO_AVC);
+
CHECK_EQ(initDecoder(), (status_t)OK);
}
@@ -100,65 +90,6 @@
delete[] mFirstPicture;
}
-void SoftAVC::initPorts() {
- OMX_PARAM_PORTDEFINITIONTYPE def;
- InitOMXParams(&def);
-
- def.nPortIndex = kInputPortIndex;
- def.eDir = OMX_DirInput;
- def.nBufferCountMin = kNumInputBuffers;
- def.nBufferCountActual = def.nBufferCountMin;
- def.nBufferSize = 8192;
- def.bEnabled = OMX_TRUE;
- def.bPopulated = OMX_FALSE;
- def.eDomain = OMX_PortDomainVideo;
- def.bBuffersContiguous = OMX_FALSE;
- def.nBufferAlignment = 1;
-
- def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_AVC);
- def.format.video.pNativeRender = NULL;
- def.format.video.nFrameWidth = mWidth;
- def.format.video.nFrameHeight = mHeight;
- def.format.video.nStride = def.format.video.nFrameWidth;
- def.format.video.nSliceHeight = def.format.video.nFrameHeight;
- def.format.video.nBitrate = 0;
- def.format.video.xFramerate = 0;
- def.format.video.bFlagErrorConcealment = OMX_FALSE;
- def.format.video.eCompressionFormat = OMX_VIDEO_CodingAVC;
- def.format.video.eColorFormat = OMX_COLOR_FormatUnused;
- def.format.video.pNativeWindow = NULL;
-
- addPort(def);
-
- def.nPortIndex = kOutputPortIndex;
- def.eDir = OMX_DirOutput;
- def.nBufferCountMin = kNumOutputBuffers;
- def.nBufferCountActual = def.nBufferCountMin;
- def.bEnabled = OMX_TRUE;
- def.bPopulated = OMX_FALSE;
- def.eDomain = OMX_PortDomainVideo;
- def.bBuffersContiguous = OMX_FALSE;
- def.nBufferAlignment = 2;
-
- def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_RAW);
- def.format.video.pNativeRender = NULL;
- def.format.video.nFrameWidth = mWidth;
- def.format.video.nFrameHeight = mHeight;
- def.format.video.nStride = def.format.video.nFrameWidth;
- def.format.video.nSliceHeight = def.format.video.nFrameHeight;
- def.format.video.nBitrate = 0;
- def.format.video.xFramerate = 0;
- def.format.video.bFlagErrorConcealment = OMX_FALSE;
- def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused;
- def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar;
- def.format.video.pNativeWindow = NULL;
-
- def.nBufferSize =
- (def.format.video.nFrameWidth * def.format.video.nFrameHeight * 3) / 2;
-
- addPort(def);
-}
-
status_t SoftAVC::initDecoder() {
// Force decoder to output buffers in display order.
if (H264SwDecInit(&mHandle, 0) == H264SWDEC_OK) {
@@ -167,126 +98,6 @@
return UNKNOWN_ERROR;
}
-OMX_ERRORTYPE SoftAVC::internalGetParameter(
- OMX_INDEXTYPE index, OMX_PTR params) {
- switch (index) {
- case OMX_IndexParamVideoPortFormat:
- {
- OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
- (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
-
- if (formatParams->nPortIndex > kOutputPortIndex) {
- return OMX_ErrorUndefined;
- }
-
- if (formatParams->nIndex != 0) {
- return OMX_ErrorNoMore;
- }
-
- if (formatParams->nPortIndex == kInputPortIndex) {
- formatParams->eCompressionFormat = OMX_VIDEO_CodingAVC;
- formatParams->eColorFormat = OMX_COLOR_FormatUnused;
- formatParams->xFramerate = 0;
- } else {
- CHECK(formatParams->nPortIndex == kOutputPortIndex);
-
- formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused;
- formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar;
- formatParams->xFramerate = 0;
- }
-
- return OMX_ErrorNone;
- }
-
- case OMX_IndexParamVideoProfileLevelQuerySupported:
- {
- OMX_VIDEO_PARAM_PROFILELEVELTYPE *profileLevel =
- (OMX_VIDEO_PARAM_PROFILELEVELTYPE *) params;
-
- if (profileLevel->nPortIndex != kInputPortIndex) {
- ALOGE("Invalid port index: %ld", profileLevel->nPortIndex);
- return OMX_ErrorUnsupportedIndex;
- }
-
- size_t index = profileLevel->nProfileIndex;
- size_t nProfileLevels =
- sizeof(kProfileLevels) / sizeof(kProfileLevels[0]);
- if (index >= nProfileLevels) {
- return OMX_ErrorNoMore;
- }
-
- profileLevel->eProfile = kProfileLevels[index].mProfile;
- profileLevel->eLevel = kProfileLevels[index].mLevel;
- return OMX_ErrorNone;
- }
-
- default:
- return SimpleSoftOMXComponent::internalGetParameter(index, params);
- }
-}
-
-OMX_ERRORTYPE SoftAVC::internalSetParameter(
- OMX_INDEXTYPE index, const OMX_PTR params) {
- switch (index) {
- case OMX_IndexParamStandardComponentRole:
- {
- const OMX_PARAM_COMPONENTROLETYPE *roleParams =
- (const OMX_PARAM_COMPONENTROLETYPE *)params;
-
- if (strncmp((const char *)roleParams->cRole,
- "video_decoder.avc",
- OMX_MAX_STRINGNAME_SIZE - 1)) {
- return OMX_ErrorUndefined;
- }
-
- return OMX_ErrorNone;
- }
-
- case OMX_IndexParamVideoPortFormat:
- {
- OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
- (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
-
- if (formatParams->nPortIndex > kOutputPortIndex) {
- return OMX_ErrorUndefined;
- }
-
- if (formatParams->nIndex != 0) {
- return OMX_ErrorNoMore;
- }
-
- return OMX_ErrorNone;
- }
-
- default:
- return SimpleSoftOMXComponent::internalSetParameter(index, params);
- }
-}
-
-OMX_ERRORTYPE SoftAVC::getConfig(
- OMX_INDEXTYPE index, OMX_PTR params) {
- switch (index) {
- case OMX_IndexConfigCommonOutputCrop:
- {
- OMX_CONFIG_RECTTYPE *rectParams = (OMX_CONFIG_RECTTYPE *)params;
-
- if (rectParams->nPortIndex != 1) {
- return OMX_ErrorUndefined;
- }
-
- rectParams->nLeft = mCropLeft;
- rectParams->nTop = mCropTop;
- rectParams->nWidth = mCropWidth;
- rectParams->nHeight = mCropHeight;
-
- return OMX_ErrorNone;
- }
-
- default:
- return OMX_ErrorUnsupportedIndex;
- }
-}
-
void SoftAVC::onQueueFilled(OMX_U32 portIndex) {
if (mSignalledError || mOutputPortSettingsChange != NONE) {
return;
@@ -298,13 +109,21 @@
List<BufferInfo *> &inQueue = getPortQueue(kInputPortIndex);
List<BufferInfo *> &outQueue = getPortQueue(kOutputPortIndex);
+
+ if (mHeadersDecoded) {
+ // Dequeue any already decoded output frames to free up space
+ // in the output queue.
+
+ drainAllOutputBuffers(false /* eos */);
+ }
+
H264SwDecRet ret = H264SWDEC_PIC_RDY;
bool portSettingsChanged = false;
while ((mEOSStatus != INPUT_DATA_AVAILABLE || !inQueue.empty())
&& outQueue.size() == kNumOutputBuffers) {
if (mEOSStatus == INPUT_EOS_SEEN) {
- drainAllOutputBuffers();
+ drainAllOutputBuffers(true /* eos */);
return;
}
@@ -392,15 +211,7 @@
mFirstPictureId = -1;
}
- while (!outQueue.empty() &&
- mHeadersDecoded &&
- H264SwDecNextPicture(mHandle, &decodedPicture, 0)
- == H264SWDEC_PIC_RDY) {
-
- int32_t picId = decodedPicture.picId;
- uint8_t *data = (uint8_t *) decodedPicture.pOutputPicture;
- drainOneOutputBuffer(picId, data);
- }
+ drainAllOutputBuffers(false /* eos */);
}
}
@@ -409,8 +220,6 @@
mWidth = info->picWidth;
mHeight = info->picHeight;
mPictureSize = mWidth * mHeight * 3 / 2;
- mCropWidth = mWidth;
- mCropHeight = mHeight;
updatePortDefinitions();
notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
mOutputPortSettingsChange = AWAITING_DISABLED;
@@ -463,43 +272,38 @@
notifyFillBufferDone(outHeader);
}
-bool SoftAVC::drainAllOutputBuffers() {
+void SoftAVC::drainAllOutputBuffers(bool eos) {
List<BufferInfo *> &outQueue = getPortQueue(kOutputPortIndex);
H264SwDecPicture decodedPicture;
+ if (mHeadersDecoded) {
+ while (!outQueue.empty()
+ && H264SWDEC_PIC_RDY == H264SwDecNextPicture(
+ mHandle, &decodedPicture, eos /* flush */)) {
+ int32_t picId = decodedPicture.picId;
+ uint8_t *data = (uint8_t *) decodedPicture.pOutputPicture;
+ drainOneOutputBuffer(picId, data);
+ }
+ }
+
+ if (!eos) {
+ return;
+ }
+
while (!outQueue.empty()) {
BufferInfo *outInfo = *outQueue.begin();
outQueue.erase(outQueue.begin());
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
- if (mHeadersDecoded &&
- H264SWDEC_PIC_RDY ==
- H264SwDecNextPicture(mHandle, &decodedPicture, 1 /* flush */)) {
- int32_t picId = decodedPicture.picId;
- CHECK(mPicToHeaderMap.indexOfKey(picId) >= 0);
-
- memcpy(outHeader->pBuffer + outHeader->nOffset,
- decodedPicture.pOutputPicture,
- mPictureSize);
-
- OMX_BUFFERHEADERTYPE *header = mPicToHeaderMap.valueFor(picId);
- outHeader->nTimeStamp = header->nTimeStamp;
- outHeader->nFlags = header->nFlags;
- outHeader->nFilledLen = mPictureSize;
- mPicToHeaderMap.removeItem(picId);
- delete header;
- } else {
- outHeader->nTimeStamp = 0;
- outHeader->nFilledLen = 0;
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
- mEOSStatus = OUTPUT_FRAMES_FLUSHED;
- }
+ outHeader->nTimeStamp = 0;
+ outHeader->nFilledLen = 0;
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
outInfo->mOwnedByUs = false;
notifyFillBufferDone(outHeader);
- }
- return true;
+ mEOSStatus = OUTPUT_FRAMES_FLUSHED;
+ }
}
void SoftAVC::onPortFlushCompleted(OMX_U32 portIndex) {
@@ -508,44 +312,9 @@
}
}
-void SoftAVC::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
- switch (mOutputPortSettingsChange) {
- case NONE:
- break;
-
- case AWAITING_DISABLED:
- {
- CHECK(!enabled);
- mOutputPortSettingsChange = AWAITING_ENABLED;
- break;
- }
-
- default:
- {
- CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
- CHECK(enabled);
- mOutputPortSettingsChange = NONE;
- break;
- }
- }
-}
-
-void SoftAVC::updatePortDefinitions() {
- OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(0)->mDef;
- def->format.video.nFrameWidth = mWidth;
- def->format.video.nFrameHeight = mHeight;
- def->format.video.nStride = def->format.video.nFrameWidth;
- def->format.video.nSliceHeight = def->format.video.nFrameHeight;
-
- def = &editPortInfo(1)->mDef;
- def->format.video.nFrameWidth = mWidth;
- def->format.video.nFrameHeight = mHeight;
- def->format.video.nStride = def->format.video.nFrameWidth;
- def->format.video.nSliceHeight = def->format.video.nFrameHeight;
-
- def->nBufferSize =
- (def->format.video.nFrameWidth
- * def->format.video.nFrameHeight * 3) / 2;
+void SoftAVC::onReset() {
+ SoftVideoDecoderOMXComponent::onReset();
+ mSignalledError = false;
}
} // namespace android
diff --git a/media/libstagefright/codecs/on2/h264dec/SoftAVC.h b/media/libstagefright/codecs/on2/h264dec/SoftAVC.h
index 879b014..ee69926 100644
--- a/media/libstagefright/codecs/on2/h264dec/SoftAVC.h
+++ b/media/libstagefright/codecs/on2/h264dec/SoftAVC.h
@@ -18,7 +18,7 @@
#define SOFT_AVC_H_
-#include "SimpleSoftOMXComponent.h"
+#include "SoftVideoDecoderOMXComponent.h"
#include <utils/KeyedVector.h>
#include "H264SwDecApi.h"
@@ -26,7 +26,7 @@
namespace android {
-struct SoftAVC : public SimpleSoftOMXComponent {
+struct SoftAVC : public SoftVideoDecoderOMXComponent {
SoftAVC(const char *name,
const OMX_CALLBACKTYPE *callbacks,
OMX_PTR appData,
@@ -35,22 +35,12 @@
protected:
virtual ~SoftAVC();
- virtual OMX_ERRORTYPE internalGetParameter(
- OMX_INDEXTYPE index, OMX_PTR params);
-
- virtual OMX_ERRORTYPE internalSetParameter(
- OMX_INDEXTYPE index, const OMX_PTR params);
-
- virtual OMX_ERRORTYPE getConfig(OMX_INDEXTYPE index, OMX_PTR params);
-
virtual void onQueueFilled(OMX_U32 portIndex);
virtual void onPortFlushCompleted(OMX_U32 portIndex);
- virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled);
+ virtual void onReset();
private:
enum {
- kInputPortIndex = 0,
- kOutputPortIndex = 1,
kNumInputBuffers = 8,
kNumOutputBuffers = 2,
};
@@ -65,9 +55,7 @@
size_t mInputBufferCount;
- uint32_t mWidth, mHeight, mPictureSize;
- uint32_t mCropLeft, mCropTop;
- uint32_t mCropWidth, mCropHeight;
+ uint32_t mPictureSize;
uint8_t *mFirstPicture;
int32_t mFirstPictureId;
@@ -81,19 +69,10 @@
EOSStatus mEOSStatus;
- enum OutputPortSettingChange {
- NONE,
- AWAITING_DISABLED,
- AWAITING_ENABLED
- };
- OutputPortSettingChange mOutputPortSettingsChange;
-
bool mSignalledError;
- void initPorts();
status_t initDecoder();
- void updatePortDefinitions();
- bool drainAllOutputBuffers();
+ void drainAllOutputBuffers(bool eos);
void drainOneOutputBuffer(int32_t picId, uint8_t *data);
void saveFirstOutputBuffer(int32_t pidId, uint8_t *data);
bool handleCropRectEvent(const CropParams* crop);
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
index 4115324..51bb958 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
@@ -424,6 +424,8 @@
delete mVi;
mVi = NULL;
}
+
+ mOutputPortSettingsChange = NONE;
}
void SoftVorbis::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
diff --git a/media/libstagefright/foundation/AHierarchicalStateMachine.cpp b/media/libstagefright/foundation/AHierarchicalStateMachine.cpp
index 40c5a3c..f7a00d8 100644
--- a/media/libstagefright/foundation/AHierarchicalStateMachine.cpp
+++ b/media/libstagefright/foundation/AHierarchicalStateMachine.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AHierarchicalStateMachine"
+#include <utils/Log.h>
+
#include <media/stagefright/foundation/AHierarchicalStateMachine.h>
#include <media/stagefright/foundation/ADebug.h>
diff --git a/media/libstagefright/httplive/Android.mk b/media/libstagefright/httplive/Android.mk
index a3fa7a3..85bd492 100644
--- a/media/libstagefright/httplive/Android.mk
+++ b/media/libstagefright/httplive/Android.mk
@@ -6,16 +6,25 @@
LiveDataSource.cpp \
LiveSession.cpp \
M3UParser.cpp \
+ PlaylistFetcher.cpp \
LOCAL_C_INCLUDES:= \
$(TOP)/frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax \
$(TOP)/external/openssl/include
+LOCAL_SHARED_LIBRARIES := \
+ libcrypto \
+ libcutils \
+ libmedia \
+ libstagefright \
+ libstagefright_foundation \
+ libutils \
+
LOCAL_MODULE:= libstagefright_httplive
ifeq ($(TARGET_ARCH),arm)
LOCAL_CFLAGS += -Wno-psabi
endif
-include $(BUILD_STATIC_LIBRARY)
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 505bdb3..e91c60b 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -18,12 +18,13 @@
#define LOG_TAG "LiveSession"
#include <utils/Log.h>
-#include "include/LiveSession.h"
+#include "LiveSession.h"
-#include "LiveDataSource.h"
+#include "M3UParser.h"
+#include "PlaylistFetcher.h"
-#include "include/M3UParser.h"
#include "include/HTTPBase.h"
+#include "mpeg2ts/AnotherPacketSource.h"
#include <cutils/properties.h>
#include <media/stagefright/foundation/hexdump.h>
@@ -33,6 +34,8 @@
#include <media/stagefright/DataSource.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
#include <ctype.h>
#include <openssl/aes.h>
@@ -47,37 +50,107 @@
mUIDValid(uidValid),
mUID(uid),
mInPreparationPhase(true),
- mDataSource(new LiveDataSource),
mHTTPDataSource(
HTTPBase::Create(
(mFlags & kFlagIncognito)
? HTTPBase::kFlagIncognito
: 0)),
mPrevBandwidthIndex(-1),
- mLastPlaylistFetchTimeUs(-1),
- mSeqNumber(-1),
- mSeekTimeUs(-1),
- mNumRetries(0),
- mStartOfPlayback(true),
- mDurationUs(-1),
- mDurationFixed(false),
- mSeekDone(false),
- mDisconnectPending(false),
- mMonitorQueueGeneration(0),
- mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY) {
+ mStreamMask(0),
+ mCheckBandwidthGeneration(0),
+ mLastDequeuedTimeUs(0ll),
+ mReconfigurationInProgress(false),
+ mDisconnectReplyID(0) {
if (mUIDValid) {
mHTTPDataSource->setUID(mUID);
}
+
+ mPacketSources.add(
+ STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */));
+
+ mPacketSources.add(
+ STREAMTYPE_VIDEO, new AnotherPacketSource(NULL /* meta */));
+
+ mPacketSources.add(
+ STREAMTYPE_SUBTITLES, new AnotherPacketSource(NULL /* meta */));
}
LiveSession::~LiveSession() {
}
-sp<DataSource> LiveSession::getDataSource() {
- return mDataSource;
+status_t LiveSession::dequeueAccessUnit(
+ StreamType stream, sp<ABuffer> *accessUnit) {
+ if (!(mStreamMask & stream)) {
+ return UNKNOWN_ERROR;
+ }
+
+ sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream);
+
+ status_t finalResult;
+ if (!packetSource->hasBufferAvailable(&finalResult)) {
+ return finalResult == OK ? -EAGAIN : finalResult;
+ }
+
+ status_t err = packetSource->dequeueAccessUnit(accessUnit);
+
+ const char *streamStr;
+ switch (stream) {
+ case STREAMTYPE_AUDIO:
+ streamStr = "audio";
+ break;
+ case STREAMTYPE_VIDEO:
+ streamStr = "video";
+ break;
+ case STREAMTYPE_SUBTITLES:
+ streamStr = "subs";
+ break;
+ default:
+ TRESPASS();
+ }
+
+ if (err == INFO_DISCONTINUITY) {
+ int32_t type;
+ CHECK((*accessUnit)->meta()->findInt32("discontinuity", &type));
+
+ sp<AMessage> extra;
+ if (!(*accessUnit)->meta()->findMessage("extra", &extra)) {
+ extra.clear();
+ }
+
+ ALOGI("[%s] read discontinuity of type %d, extra = %s",
+ streamStr,
+ type,
+ extra == NULL ? "NULL" : extra->debugString().c_str());
+ } else if (err == OK) {
+ int64_t timeUs;
+ CHECK((*accessUnit)->meta()->findInt64("timeUs", &timeUs));
+ ALOGV("[%s] read buffer at time %lld us", streamStr, timeUs);
+
+ mLastDequeuedTimeUs = timeUs;
+ } else {
+ ALOGI("[%s] encountered error %d", streamStr, err);
+ }
+
+ return err;
}
-void LiveSession::connect(
+status_t LiveSession::getStreamFormat(StreamType stream, sp<AMessage> *format) {
+ if (!(mStreamMask & stream)) {
+ return UNKNOWN_ERROR;
+ }
+
+ sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream);
+
+ sp<MetaData> meta = packetSource->getFormat();
+
+ if (meta == NULL) {
+ return -EAGAIN;
+ }
+
+ return convertMetaDataToMessage(meta, format);
+}
+
+void LiveSession::connectAsync(
const char *url, const KeyedVector<String8, String8> *headers) {
sp<AMessage> msg = new AMessage(kWhatConnect, id());
msg->setString("url", url);
@@ -91,55 +164,184 @@
msg->post();
}
-void LiveSession::disconnect() {
- Mutex::Autolock autoLock(mLock);
- mDisconnectPending = true;
+status_t LiveSession::disconnect() {
+ sp<AMessage> msg = new AMessage(kWhatDisconnect, id());
- mHTTPDataSource->disconnect();
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
- (new AMessage(kWhatDisconnect, id()))->post();
+ return err;
}
-void LiveSession::seekTo(int64_t timeUs) {
- Mutex::Autolock autoLock(mLock);
- mSeekDone = false;
-
+status_t LiveSession::seekTo(int64_t timeUs) {
sp<AMessage> msg = new AMessage(kWhatSeek, id());
msg->setInt64("timeUs", timeUs);
- msg->post();
- while (!mSeekDone) {
- mCondition.wait(mLock);
- }
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+
+ return err;
}
void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
switch (msg->what()) {
case kWhatConnect:
+ {
onConnect(msg);
break;
+ }
case kWhatDisconnect:
- onDisconnect();
- break;
-
- case kWhatMonitorQueue:
{
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
+ CHECK(msg->senderAwaitsResponse(&mDisconnectReplyID));
- if (generation != mMonitorQueueGeneration) {
- // Stale event
+ if (mReconfigurationInProgress) {
break;
}
- onMonitorQueue();
+ finishDisconnect();
break;
}
case kWhatSeek:
- onSeek(msg);
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ status_t err = onSeek(msg);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+
+ response->postReply(replyID);
break;
+ }
+
+ case kWhatFetcherNotify:
+ {
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case PlaylistFetcher::kWhatStarted:
+ break;
+ case PlaylistFetcher::kWhatPaused:
+ case PlaylistFetcher::kWhatStopped:
+ {
+ if (what == PlaylistFetcher::kWhatStopped) {
+ AString uri;
+ CHECK(msg->findString("uri", &uri));
+ mFetcherInfos.removeItem(uri);
+ }
+
+ if (mContinuation != NULL) {
+ CHECK_GT(mContinuationCounter, 0);
+ if (--mContinuationCounter == 0) {
+ mContinuation->post();
+ }
+ }
+ break;
+ }
+
+ case PlaylistFetcher::kWhatDurationUpdate:
+ {
+ AString uri;
+ CHECK(msg->findString("uri", &uri));
+
+ int64_t durationUs;
+ CHECK(msg->findInt64("durationUs", &durationUs));
+
+ FetcherInfo *info = &mFetcherInfos.editValueFor(uri);
+ info->mDurationUs = durationUs;
+ break;
+ }
+
+ case PlaylistFetcher::kWhatError:
+ {
+ status_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ ALOGE("XXX Received error %d from PlaylistFetcher.", err);
+
+ if (mInPreparationPhase) {
+ postPrepared(err);
+ }
+
+ mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(err);
+
+ mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(err);
+
+ mPacketSources.valueFor(
+ STREAMTYPE_SUBTITLES)->signalEOS(err);
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatError);
+ notify->setInt32("err", err);
+ notify->post();
+ break;
+ }
+
+ case PlaylistFetcher::kWhatTemporarilyDoneFetching:
+ {
+ AString uri;
+ CHECK(msg->findString("uri", &uri));
+
+ FetcherInfo *info = &mFetcherInfos.editValueFor(uri);
+ info->mIsPrepared = true;
+
+ if (mInPreparationPhase) {
+ bool allFetchersPrepared = true;
+ for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ if (!mFetcherInfos.valueAt(i).mIsPrepared) {
+ allFetchersPrepared = false;
+ break;
+ }
+ }
+
+ if (allFetchersPrepared) {
+ postPrepared(OK);
+ }
+ }
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+
+ break;
+ }
+
+ case kWhatCheckBandwidth:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mCheckBandwidthGeneration) {
+ break;
+ }
+
+ onCheckBandwidth();
+ break;
+ }
+
+ case kWhatChangeConfiguration2:
+ {
+ onChangeConfiguration2(msg);
+ break;
+ }
+
+ case kWhatChangeConfiguration3:
+ {
+ onChangeConfiguration3(msg);
+ break;
+ }
+
+ case kWhatFinishDisconnect2:
+ {
+ onFinishDisconnect2();
+ break;
+ }
default:
TRESPASS();
@@ -172,48 +374,127 @@
headers = NULL;
}
+#if 1
ALOGI("onConnect <URL suppressed>");
+#else
+ ALOGI("onConnect %s", url.c_str());
+#endif
mMasterURL = url;
bool dummy;
- sp<M3UParser> playlist = fetchPlaylist(url.c_str(), &dummy);
+ mPlaylist = fetchPlaylist(url.c_str(), NULL /* curPlaylistHash */, &dummy);
- if (playlist == NULL) {
+ if (mPlaylist == NULL) {
ALOGE("unable to fetch master playlist '%s'.", url.c_str());
- signalEOS(ERROR_IO);
+ postPrepared(ERROR_IO);
return;
}
- if (playlist->isVariantPlaylist()) {
- for (size_t i = 0; i < playlist->size(); ++i) {
+ // We trust the content provider to make a reasonable choice of preferred
+ // initial bandwidth by listing it first in the variant playlist.
+ // At startup we really don't have a good estimate on the available
+ // network bandwidth since we haven't tranferred any data yet. Once
+ // we have we can make a better informed choice.
+ size_t initialBandwidth = 0;
+ size_t initialBandwidthIndex = 0;
+
+ if (mPlaylist->isVariantPlaylist()) {
+ for (size_t i = 0; i < mPlaylist->size(); ++i) {
BandwidthItem item;
+ item.mPlaylistIndex = i;
+
sp<AMessage> meta;
- playlist->itemAt(i, &item.mURI, &meta);
+ AString uri;
+ mPlaylist->itemAt(i, &uri, &meta);
unsigned long bandwidth;
CHECK(meta->findInt32("bandwidth", (int32_t *)&item.mBandwidth));
+ if (initialBandwidth == 0) {
+ initialBandwidth = item.mBandwidth;
+ }
+
mBandwidthItems.push(item);
}
CHECK_GT(mBandwidthItems.size(), 0u);
mBandwidthItems.sort(SortByBandwidth);
+
+ for (size_t i = 0; i < mBandwidthItems.size(); ++i) {
+ if (mBandwidthItems.itemAt(i).mBandwidth == initialBandwidth) {
+ initialBandwidthIndex = i;
+ break;
+ }
+ }
+ } else {
+ // dummy item.
+ BandwidthItem item;
+ item.mPlaylistIndex = 0;
+ item.mBandwidth = 0;
+ mBandwidthItems.push(item);
}
- postMonitorQueue();
+ changeConfiguration(0ll /* timeUs */, initialBandwidthIndex);
}
-void LiveSession::onDisconnect() {
- ALOGI("onDisconnect");
+void LiveSession::finishDisconnect() {
+ // No reconfiguration is currently pending, make sure none will trigger
+ // during disconnection either.
+ cancelCheckBandwidthEvent();
- signalEOS(ERROR_END_OF_STREAM);
+ for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ mFetcherInfos.valueAt(i).mFetcher->stopAsync();
+ }
- Mutex::Autolock autoLock(mLock);
- mDisconnectPending = false;
+ sp<AMessage> msg = new AMessage(kWhatFinishDisconnect2, id());
+
+ mContinuationCounter = mFetcherInfos.size();
+ mContinuation = msg;
+
+ if (mContinuationCounter == 0) {
+ msg->post();
+ }
+}
+
+void LiveSession::onFinishDisconnect2() {
+ mContinuation.clear();
+
+ mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(ERROR_END_OF_STREAM);
+ mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(ERROR_END_OF_STREAM);
+
+ mPacketSources.valueFor(
+ STREAMTYPE_SUBTITLES)->signalEOS(ERROR_END_OF_STREAM);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", OK);
+
+ response->postReply(mDisconnectReplyID);
+ mDisconnectReplyID = 0;
+}
+
+sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) {
+ ssize_t index = mFetcherInfos.indexOfKey(uri);
+
+ if (index >= 0) {
+ return NULL;
+ }
+
+ sp<AMessage> notify = new AMessage(kWhatFetcherNotify, id());
+ notify->setString("uri", uri);
+
+ FetcherInfo info;
+ info.mFetcher = new PlaylistFetcher(notify, this, uri);
+ info.mDurationUs = -1ll;
+ info.mIsPrepared = false;
+ looper()->registerHandler(info.mFetcher);
+
+ mFetcherInfos.add(uri, info);
+
+ return info.mFetcher;
}
status_t LiveSession::fetchFile(
@@ -229,14 +510,6 @@
&& strncasecmp(url, "https://", 8)) {
return ERROR_UNSUPPORTED;
} else {
- {
- Mutex::Autolock autoLock(mLock);
-
- if (mDisconnectPending) {
- return ERROR_IO;
- }
- }
-
KeyedVector<String8, String8> headers = mExtraHeaders;
if (range_offset > 0 || range_length >= 0) {
headers.add(
@@ -315,7 +588,8 @@
return OK;
}
-sp<M3UParser> LiveSession::fetchPlaylist(const char *url, bool *unchanged) {
+sp<M3UParser> LiveSession::fetchPlaylist(
+ const char *url, uint8_t *curPlaylistHash, bool *unchanged) {
ALOGV("fetchPlaylist '%s'", url);
*unchanged = false;
@@ -339,13 +613,8 @@
MD5_Final(hash, &m);
- if (mPlaylist != NULL && !memcmp(hash, mPlaylistHash, 16)) {
+ if (curPlaylistHash != NULL && !memcmp(hash, curPlaylistHash, 16)) {
// playlist unchanged
-
- if (mRefreshState != THIRD_UNCHANGED_RELOAD_ATTEMPT) {
- mRefreshState = (RefreshState)(mRefreshState + 1);
- }
-
*unchanged = true;
ALOGV("Playlist unchanged, refresh state is now %d",
@@ -354,9 +623,9 @@
return NULL;
}
- memcpy(mPlaylistHash, hash, sizeof(hash));
-
- mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY;
+ if (curPlaylistHash != NULL) {
+ memcpy(curPlaylistHash, hash, sizeof(hash));
+ }
#endif
sp<M3UParser> playlist =
@@ -371,37 +640,6 @@
return playlist;
}
-int64_t LiveSession::getSegmentStartTimeUs(int32_t seqNumber) const {
- CHECK(mPlaylist != NULL);
-
- int32_t firstSeqNumberInPlaylist;
- if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
- "media-sequence", &firstSeqNumberInPlaylist)) {
- firstSeqNumberInPlaylist = 0;
- }
-
- int32_t lastSeqNumberInPlaylist =
- firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
-
- CHECK_GE(seqNumber, firstSeqNumberInPlaylist);
- CHECK_LE(seqNumber, lastSeqNumberInPlaylist);
-
- int64_t segmentStartUs = 0ll;
- for (int32_t index = 0;
- index < seqNumber - firstSeqNumberInPlaylist; ++index) {
- sp<AMessage> itemMeta;
- CHECK(mPlaylist->itemAt(
- index, NULL /* uri */, &itemMeta));
-
- int64_t itemDurationUs;
- CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
-
- segmentStartUs += itemDurationUs;
- }
-
- return segmentStartUs;
-}
-
static double uniformRand() {
return (double)rand() / RAND_MAX;
}
@@ -412,36 +650,50 @@
}
#if 1
- int32_t bandwidthBps;
- if (mHTTPDataSource != NULL
- && mHTTPDataSource->estimateBandwidth(&bandwidthBps)) {
- ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f);
- } else {
- ALOGV("no bandwidth estimate.");
- return 0; // Pick the lowest bandwidth stream by default.
- }
-
char value[PROPERTY_VALUE_MAX];
- if (property_get("media.httplive.max-bw", value, NULL)) {
+ ssize_t index = -1;
+ if (property_get("media.httplive.bw-index", value, NULL)) {
char *end;
- long maxBw = strtoul(value, &end, 10);
- if (end > value && *end == '\0') {
- if (maxBw > 0 && bandwidthBps > maxBw) {
- ALOGV("bandwidth capped to %ld bps", maxBw);
- bandwidthBps = maxBw;
- }
+ index = strtol(value, &end, 10);
+ CHECK(end > value && *end == '\0');
+
+ if (index >= 0 && (size_t)index >= mBandwidthItems.size()) {
+ index = mBandwidthItems.size() - 1;
}
}
- // Consider only 80% of the available bandwidth usable.
- bandwidthBps = (bandwidthBps * 8) / 10;
+ if (index < 0) {
+ int32_t bandwidthBps;
+ if (mHTTPDataSource != NULL
+ && mHTTPDataSource->estimateBandwidth(&bandwidthBps)) {
+ ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f);
+ } else {
+ ALOGV("no bandwidth estimate.");
+ return 0; // Pick the lowest bandwidth stream by default.
+ }
- // Pick the highest bandwidth stream below or equal to estimated bandwidth.
+ char value[PROPERTY_VALUE_MAX];
+ if (property_get("media.httplive.max-bw", value, NULL)) {
+ char *end;
+ long maxBw = strtoul(value, &end, 10);
+ if (end > value && *end == '\0') {
+ if (maxBw > 0 && bandwidthBps > maxBw) {
+ ALOGV("bandwidth capped to %ld bps", maxBw);
+ bandwidthBps = maxBw;
+ }
+ }
+ }
- size_t index = mBandwidthItems.size() - 1;
- while (index > 0 && mBandwidthItems.itemAt(index).mBandwidth
- > (size_t)bandwidthBps) {
- --index;
+ // Consider only 80% of the available bandwidth usable.
+ bandwidthBps = (bandwidthBps * 8) / 10;
+
+ // Pick the highest bandwidth stream below or equal to estimated bandwidth.
+
+ index = mBandwidthItems.size() - 1;
+ while (index > 0 && mBandwidthItems.itemAt(index).mBandwidth
+ > (size_t)bandwidthBps) {
+ --index;
+ }
}
#elif 0
// Change bandwidth at random()
@@ -452,6 +704,8 @@
// to lowest)
const size_t kMinIndex = 0;
+ static ssize_t mPrevBandwidthIndex = -1;
+
size_t index;
if (mPrevBandwidthIndex < 0) {
index = kMinIndex;
@@ -463,6 +717,7 @@
index = kMinIndex;
}
}
+ mPrevBandwidthIndex = index;
#elif 0
// Pick the highest bandwidth stream below or equal to 1.2 Mbit/sec
@@ -470,559 +725,51 @@
while (index > 0 && mBandwidthItems.itemAt(index).mBandwidth > 1200000) {
--index;
}
+#elif 1
+ char value[PROPERTY_VALUE_MAX];
+ size_t index;
+ if (property_get("media.httplive.bw-index", value, NULL)) {
+ char *end;
+ index = strtoul(value, &end, 10);
+ CHECK(end > value && *end == '\0');
+
+ if (index >= mBandwidthItems.size()) {
+ index = mBandwidthItems.size() - 1;
+ }
+ } else {
+ index = 0;
+ }
#else
size_t index = mBandwidthItems.size() - 1; // Highest bandwidth stream
#endif
+ CHECK_GE(index, 0);
+
return index;
}
-bool LiveSession::timeToRefreshPlaylist(int64_t nowUs) const {
- if (mPlaylist == NULL) {
- CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY);
- return true;
+status_t LiveSession::onSeek(const sp<AMessage> &msg) {
+ int64_t timeUs;
+ CHECK(msg->findInt64("timeUs", &timeUs));
+
+ if (!mReconfigurationInProgress) {
+ changeConfiguration(timeUs, getBandwidthIndex());
}
- int32_t targetDurationSecs;
- CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
-
- int64_t targetDurationUs = targetDurationSecs * 1000000ll;
-
- int64_t minPlaylistAgeUs;
-
- switch (mRefreshState) {
- case INITIAL_MINIMUM_RELOAD_DELAY:
- {
- size_t n = mPlaylist->size();
- if (n > 0) {
- sp<AMessage> itemMeta;
- CHECK(mPlaylist->itemAt(n - 1, NULL /* uri */, &itemMeta));
-
- int64_t itemDurationUs;
- CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
-
- minPlaylistAgeUs = itemDurationUs;
- break;
- }
-
- // fall through
- }
-
- case FIRST_UNCHANGED_RELOAD_ATTEMPT:
- {
- minPlaylistAgeUs = targetDurationUs / 2;
- break;
- }
-
- case SECOND_UNCHANGED_RELOAD_ATTEMPT:
- {
- minPlaylistAgeUs = (targetDurationUs * 3) / 2;
- break;
- }
-
- case THIRD_UNCHANGED_RELOAD_ATTEMPT:
- {
- minPlaylistAgeUs = targetDurationUs * 3;
- break;
- }
-
- default:
- TRESPASS();
- break;
- }
-
- return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs;
-}
-
-void LiveSession::onDownloadNext() {
- size_t bandwidthIndex = getBandwidthIndex();
-
-rinse_repeat:
- int64_t nowUs = ALooper::GetNowUs();
-
- if (mLastPlaylistFetchTimeUs < 0
- || (ssize_t)bandwidthIndex != mPrevBandwidthIndex
- || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) {
- AString url;
- if (mBandwidthItems.size() > 0) {
- url = mBandwidthItems.editItemAt(bandwidthIndex).mURI;
- } else {
- url = mMasterURL;
- }
-
- if ((ssize_t)bandwidthIndex != mPrevBandwidthIndex) {
- // If we switch bandwidths, do not pay any heed to whether
- // playlists changed since the last time...
- mPlaylist.clear();
- }
-
- bool unchanged;
- sp<M3UParser> playlist = fetchPlaylist(url.c_str(), &unchanged);
- if (playlist == NULL) {
- if (unchanged) {
- // We succeeded in fetching the playlist, but it was
- // unchanged from the last time we tried.
- } else {
- ALOGE("failed to load playlist at url '%s'", url.c_str());
- signalEOS(ERROR_IO);
-
- return;
- }
- } else {
- mPlaylist = playlist;
- }
-
- if (!mDurationFixed) {
- Mutex::Autolock autoLock(mLock);
-
- if (!mPlaylist->isComplete() && !mPlaylist->isEvent()) {
- mDurationUs = -1;
- mDurationFixed = true;
- } else {
- mDurationUs = 0;
- for (size_t i = 0; i < mPlaylist->size(); ++i) {
- sp<AMessage> itemMeta;
- CHECK(mPlaylist->itemAt(
- i, NULL /* uri */, &itemMeta));
-
- int64_t itemDurationUs;
- CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
-
- mDurationUs += itemDurationUs;
- }
-
- mDurationFixed = mPlaylist->isComplete();
- }
- }
-
- mLastPlaylistFetchTimeUs = ALooper::GetNowUs();
- }
-
- int32_t firstSeqNumberInPlaylist;
- if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
- "media-sequence", &firstSeqNumberInPlaylist)) {
- firstSeqNumberInPlaylist = 0;
- }
-
- bool seekDiscontinuity = false;
- bool explicitDiscontinuity = false;
- bool bandwidthChanged = false;
-
- if (mSeekTimeUs >= 0) {
- if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
- size_t index = 0;
- int64_t segmentStartUs = 0;
- while (index < mPlaylist->size()) {
- sp<AMessage> itemMeta;
- CHECK(mPlaylist->itemAt(
- index, NULL /* uri */, &itemMeta));
-
- int64_t itemDurationUs;
- CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
-
- if (mSeekTimeUs < segmentStartUs + itemDurationUs) {
- break;
- }
-
- segmentStartUs += itemDurationUs;
- ++index;
- }
-
- if (index < mPlaylist->size()) {
- int32_t newSeqNumber = firstSeqNumberInPlaylist + index;
-
- ALOGI("seeking to seq no %d", newSeqNumber);
-
- mSeqNumber = newSeqNumber;
-
- mDataSource->reset();
-
- // reseting the data source will have had the
- // side effect of discarding any previously queued
- // bandwidth change discontinuity.
- // Therefore we'll need to treat these seek
- // discontinuities as involving a bandwidth change
- // even if they aren't directly.
- seekDiscontinuity = true;
- bandwidthChanged = true;
- }
- }
-
- mSeekTimeUs = -1;
-
- Mutex::Autolock autoLock(mLock);
- mSeekDone = true;
- mCondition.broadcast();
- }
-
- const int32_t lastSeqNumberInPlaylist =
- firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
-
- if (mSeqNumber < 0) {
- if (mPlaylist->isComplete()) {
- mSeqNumber = firstSeqNumberInPlaylist;
- } else {
- // If this is a live session, start 3 segments from the end.
- mSeqNumber = lastSeqNumberInPlaylist - 3;
- if (mSeqNumber < firstSeqNumberInPlaylist) {
- mSeqNumber = firstSeqNumberInPlaylist;
- }
- }
- }
-
- if (mSeqNumber < firstSeqNumberInPlaylist
- || mSeqNumber > lastSeqNumberInPlaylist) {
- if (mPrevBandwidthIndex != (ssize_t)bandwidthIndex) {
- // Go back to the previous bandwidth.
-
- ALOGI("new bandwidth does not have the sequence number "
- "we're looking for, switching back to previous bandwidth");
-
- mLastPlaylistFetchTimeUs = -1;
- bandwidthIndex = mPrevBandwidthIndex;
- goto rinse_repeat;
- }
-
- if (!mPlaylist->isComplete() && mNumRetries < kMaxNumRetries) {
- ++mNumRetries;
-
- if (mSeqNumber > lastSeqNumberInPlaylist) {
- mLastPlaylistFetchTimeUs = -1;
- postMonitorQueue(3000000ll);
- return;
- }
-
- // we've missed the boat, let's start from the lowest sequence
- // number available and signal a discontinuity.
-
- ALOGI("We've missed the boat, restarting playback.");
- mSeqNumber = lastSeqNumberInPlaylist;
- explicitDiscontinuity = true;
-
- // fall through
- } else {
- ALOGE("Cannot find sequence number %d in playlist "
- "(contains %d - %d)",
- mSeqNumber, firstSeqNumberInPlaylist,
- firstSeqNumberInPlaylist + mPlaylist->size() - 1);
-
- signalEOS(ERROR_END_OF_STREAM);
- return;
- }
- }
-
- mNumRetries = 0;
-
- AString uri;
- sp<AMessage> itemMeta;
- CHECK(mPlaylist->itemAt(
- mSeqNumber - firstSeqNumberInPlaylist,
- &uri,
- &itemMeta));
-
- int32_t val;
- if (itemMeta->findInt32("discontinuity", &val) && val != 0) {
- explicitDiscontinuity = true;
- }
-
- int64_t range_offset, range_length;
- if (!itemMeta->findInt64("range-offset", &range_offset)
- || !itemMeta->findInt64("range-length", &range_length)) {
- range_offset = 0;
- range_length = -1;
- }
-
- ALOGV("fetching segment %d from (%d .. %d)",
- mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist);
-
- sp<ABuffer> buffer;
- status_t err = fetchFile(uri.c_str(), &buffer, range_offset, range_length);
- if (err != OK) {
- ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());
- signalEOS(err);
- return;
- }
-
- CHECK(buffer != NULL);
-
- err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer);
-
- if (err != OK) {
- ALOGE("decryptBuffer failed w/ error %d", err);
-
- signalEOS(err);
- return;
- }
-
- if (buffer->size() == 0 || buffer->data()[0] != 0x47) {
- // Not a transport stream???
-
- ALOGE("This doesn't look like a transport stream...");
-
- mBandwidthItems.removeAt(bandwidthIndex);
-
- if (mBandwidthItems.isEmpty()) {
- signalEOS(ERROR_UNSUPPORTED);
- return;
- }
-
- ALOGI("Retrying with a different bandwidth stream.");
-
- mLastPlaylistFetchTimeUs = -1;
- bandwidthIndex = getBandwidthIndex();
- mPrevBandwidthIndex = bandwidthIndex;
- mSeqNumber = -1;
-
- goto rinse_repeat;
- }
-
- if ((size_t)mPrevBandwidthIndex != bandwidthIndex) {
- bandwidthChanged = true;
- }
-
- if (mPrevBandwidthIndex < 0) {
- // Don't signal a bandwidth change at the very beginning of
- // playback.
- bandwidthChanged = false;
- }
-
- if (mStartOfPlayback) {
- seekDiscontinuity = true;
- mStartOfPlayback = false;
- }
-
- if (seekDiscontinuity || explicitDiscontinuity || bandwidthChanged) {
- // Signal discontinuity.
-
- ALOGI("queueing discontinuity (seek=%d, explicit=%d, bandwidthChanged=%d)",
- seekDiscontinuity, explicitDiscontinuity, bandwidthChanged);
-
- sp<ABuffer> tmp = new ABuffer(188);
- memset(tmp->data(), 0, tmp->size());
-
- // signal a 'hard' discontinuity for explicit or bandwidthChanged.
- uint8_t type = (explicitDiscontinuity || bandwidthChanged) ? 1 : 0;
-
- if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
- // If this was a live event this made no sense since
- // we don't have access to all the segment before the current
- // one.
- int64_t segmentStartTimeUs = getSegmentStartTimeUs(mSeqNumber);
- memcpy(tmp->data() + 2, &segmentStartTimeUs, sizeof(segmentStartTimeUs));
-
- type |= 2;
- }
-
- tmp->data()[1] = type;
-
- mDataSource->queueBuffer(tmp);
- }
-
- mDataSource->queueBuffer(buffer);
-
- mPrevBandwidthIndex = bandwidthIndex;
- ++mSeqNumber;
-
- postMonitorQueue();
-}
-
-void LiveSession::signalEOS(status_t err) {
- if (mInPreparationPhase && mNotify != NULL) {
- sp<AMessage> notify = mNotify->dup();
-
- notify->setInt32(
- "what",
- err == ERROR_END_OF_STREAM
- ? kWhatPrepared : kWhatPreparationFailed);
-
- if (err != ERROR_END_OF_STREAM) {
- notify->setInt32("err", err);
- }
-
- notify->post();
-
- mInPreparationPhase = false;
- }
-
- mDataSource->queueEOS(err);
-}
-
-void LiveSession::onMonitorQueue() {
- if (mSeekTimeUs >= 0
- || mDataSource->countQueuedBuffers() < kMaxNumQueuedFragments) {
- onDownloadNext();
- } else {
- if (mInPreparationPhase) {
- if (mNotify != NULL) {
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatPrepared);
- notify->post();
- }
-
- mInPreparationPhase = false;
- }
-
- postMonitorQueue(1000000ll);
- }
-}
-
-status_t LiveSession::decryptBuffer(
- size_t playlistIndex, const sp<ABuffer> &buffer) {
- sp<AMessage> itemMeta;
- bool found = false;
- AString method;
-
- for (ssize_t i = playlistIndex; i >= 0; --i) {
- AString uri;
- CHECK(mPlaylist->itemAt(i, &uri, &itemMeta));
-
- if (itemMeta->findString("cipher-method", &method)) {
- found = true;
- break;
- }
- }
-
- if (!found) {
- method = "NONE";
- }
-
- if (method == "NONE") {
- return OK;
- } else if (!(method == "AES-128")) {
- ALOGE("Unsupported cipher method '%s'", method.c_str());
- return ERROR_UNSUPPORTED;
- }
-
- AString keyURI;
- if (!itemMeta->findString("cipher-uri", &keyURI)) {
- ALOGE("Missing key uri");
- return ERROR_MALFORMED;
- }
-
- ssize_t index = mAESKeyForURI.indexOfKey(keyURI);
-
- sp<ABuffer> key;
- if (index >= 0) {
- key = mAESKeyForURI.valueAt(index);
- } else {
- key = new ABuffer(16);
-
- sp<HTTPBase> keySource =
- HTTPBase::Create(
- (mFlags & kFlagIncognito)
- ? HTTPBase::kFlagIncognito
- : 0);
-
- if (mUIDValid) {
- keySource->setUID(mUID);
- }
-
- status_t err =
- keySource->connect(
- keyURI.c_str(),
- mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
-
- if (err == OK) {
- size_t offset = 0;
- while (offset < 16) {
- ssize_t n = keySource->readAt(
- offset, key->data() + offset, 16 - offset);
- if (n <= 0) {
- err = ERROR_IO;
- break;
- }
-
- offset += n;
- }
- }
-
- if (err != OK) {
- ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str());
- return ERROR_IO;
- }
-
- mAESKeyForURI.add(keyURI, key);
- }
-
- AES_KEY aes_key;
- if (AES_set_decrypt_key(key->data(), 128, &aes_key) != 0) {
- ALOGE("failed to set AES decryption key.");
- return UNKNOWN_ERROR;
- }
-
- unsigned char aes_ivec[16];
-
- AString iv;
- if (itemMeta->findString("cipher-iv", &iv)) {
- if ((!iv.startsWith("0x") && !iv.startsWith("0X"))
- || iv.size() != 16 * 2 + 2) {
- ALOGE("malformed cipher IV '%s'.", iv.c_str());
- return ERROR_MALFORMED;
- }
-
- memset(aes_ivec, 0, sizeof(aes_ivec));
- for (size_t i = 0; i < 16; ++i) {
- char c1 = tolower(iv.c_str()[2 + 2 * i]);
- char c2 = tolower(iv.c_str()[3 + 2 * i]);
- if (!isxdigit(c1) || !isxdigit(c2)) {
- ALOGE("malformed cipher IV '%s'.", iv.c_str());
- return ERROR_MALFORMED;
- }
- uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10;
- uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10;
-
- aes_ivec[i] = nibble1 << 4 | nibble2;
- }
- } else {
- memset(aes_ivec, 0, sizeof(aes_ivec));
- aes_ivec[15] = mSeqNumber & 0xff;
- aes_ivec[14] = (mSeqNumber >> 8) & 0xff;
- aes_ivec[13] = (mSeqNumber >> 16) & 0xff;
- aes_ivec[12] = (mSeqNumber >> 24) & 0xff;
- }
-
- AES_cbc_encrypt(
- buffer->data(), buffer->data(), buffer->size(),
- &aes_key, aes_ivec, AES_DECRYPT);
-
- // hexdump(buffer->data(), buffer->size());
-
- size_t n = buffer->size();
- CHECK_GT(n, 0u);
-
- size_t pad = buffer->data()[n - 1];
-
- CHECK_GT(pad, 0u);
- CHECK_LE(pad, 16u);
- CHECK_GE((size_t)n, pad);
- for (size_t i = 0; i < pad; ++i) {
- CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad);
- }
-
- n -= pad;
-
- buffer->setRange(buffer->offset(), n);
-
return OK;
}
-void LiveSession::postMonitorQueue(int64_t delayUs) {
- sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id());
- msg->setInt32("generation", ++mMonitorQueueGeneration);
- msg->post(delayUs);
-}
-
-void LiveSession::onSeek(const sp<AMessage> &msg) {
- int64_t timeUs;
- CHECK(msg->findInt64("timeUs", &timeUs));
-
- mSeekTimeUs = timeUs;
- postMonitorQueue();
-}
-
status_t LiveSession::getDuration(int64_t *durationUs) const {
- Mutex::Autolock autoLock(mLock);
- *durationUs = mDurationUs;
+ int64_t maxDurationUs = 0ll;
+ for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ int64_t fetcherDurationUs = mFetcherInfos.valueAt(i).mDurationUs;
+
+ if (fetcherDurationUs >= 0ll && fetcherDurationUs > maxDurationUs) {
+ maxDurationUs = fetcherDurationUs;
+ }
+ }
+
+ *durationUs = maxDurationUs;
return OK;
}
@@ -1033,7 +780,326 @@
}
bool LiveSession::hasDynamicDuration() const {
- return !mDurationFixed;
+ return false;
+}
+
+void LiveSession::changeConfiguration(int64_t timeUs, size_t bandwidthIndex) {
+ CHECK(!mReconfigurationInProgress);
+ mReconfigurationInProgress = true;
+
+ mPrevBandwidthIndex = bandwidthIndex;
+
+ ALOGV("changeConfiguration => timeUs:%lld us, bwIndex:%d",
+ timeUs, bandwidthIndex);
+
+ mPlaylist->pickRandomMediaItems();
+
+ CHECK_LT(bandwidthIndex, mBandwidthItems.size());
+ const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex);
+
+ uint32_t streamMask = 0;
+
+ AString audioURI;
+ if (mPlaylist->getAudioURI(item.mPlaylistIndex, &audioURI)) {
+ streamMask |= STREAMTYPE_AUDIO;
+ }
+
+ AString videoURI;
+ if (mPlaylist->getVideoURI(item.mPlaylistIndex, &videoURI)) {
+ streamMask |= STREAMTYPE_VIDEO;
+ }
+
+ AString subtitleURI;
+ if (mPlaylist->getSubtitleURI(item.mPlaylistIndex, &subtitleURI)) {
+ streamMask |= STREAMTYPE_SUBTITLES;
+ }
+
+ // Step 1, stop and discard fetchers that are no longer needed.
+ // Pause those that we'll reuse.
+ for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ const AString &uri = mFetcherInfos.keyAt(i);
+
+ bool discardFetcher = true;
+
+ // If we're seeking all current fetchers are discarded.
+ if (timeUs < 0ll) {
+ if (((streamMask & STREAMTYPE_AUDIO) && uri == audioURI)
+ || ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI)
+ || ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI)) {
+ discardFetcher = false;
+ }
+ }
+
+ if (discardFetcher) {
+ mFetcherInfos.valueAt(i).mFetcher->stopAsync();
+ } else {
+ mFetcherInfos.valueAt(i).mFetcher->pauseAsync();
+ }
+ }
+
+ sp<AMessage> msg = new AMessage(kWhatChangeConfiguration2, id());
+ msg->setInt32("streamMask", streamMask);
+ msg->setInt64("timeUs", timeUs);
+ if (streamMask & STREAMTYPE_AUDIO) {
+ msg->setString("audioURI", audioURI.c_str());
+ }
+ if (streamMask & STREAMTYPE_VIDEO) {
+ msg->setString("videoURI", videoURI.c_str());
+ }
+ if (streamMask & STREAMTYPE_SUBTITLES) {
+ msg->setString("subtitleURI", subtitleURI.c_str());
+ }
+
+ // Every time a fetcher acknowledges the stopAsync or pauseAsync request
+ // we'll decrement mContinuationCounter, once it reaches zero, i.e. all
+ // fetchers have completed their asynchronous operation, we'll post
+ // mContinuation, which then is handled below in onChangeConfiguration2.
+ mContinuationCounter = mFetcherInfos.size();
+ mContinuation = msg;
+
+ if (mContinuationCounter == 0) {
+ msg->post();
+ }
+}
+
+void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) {
+ mContinuation.clear();
+
+ // All fetchers are either suspended or have been removed now.
+
+ uint32_t streamMask;
+ CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
+
+ AString audioURI, videoURI, subtitleURI;
+ if (streamMask & STREAMTYPE_AUDIO) {
+ CHECK(msg->findString("audioURI", &audioURI));
+ ALOGV("audioURI = '%s'", audioURI.c_str());
+ }
+ if (streamMask & STREAMTYPE_VIDEO) {
+ CHECK(msg->findString("videoURI", &videoURI));
+ ALOGV("videoURI = '%s'", videoURI.c_str());
+ }
+ if (streamMask & STREAMTYPE_SUBTITLES) {
+ CHECK(msg->findString("subtitleURI", &subtitleURI));
+ ALOGV("subtitleURI = '%s'", subtitleURI.c_str());
+ }
+
+ // Determine which decoders to shutdown on the player side,
+ // a decoder has to be shutdown if either
+ // 1) its streamtype was active before but now longer isn't.
+ // or
+ // 2) its streamtype was already active and still is but the URI
+ // has changed.
+ uint32_t changedMask = 0;
+ if (((mStreamMask & streamMask & STREAMTYPE_AUDIO)
+ && !(audioURI == mAudioURI))
+ || (mStreamMask & ~streamMask & STREAMTYPE_AUDIO)) {
+ changedMask |= STREAMTYPE_AUDIO;
+ }
+ if (((mStreamMask & streamMask & STREAMTYPE_VIDEO)
+ && !(videoURI == mVideoURI))
+ || (mStreamMask & ~streamMask & STREAMTYPE_VIDEO)) {
+ changedMask |= STREAMTYPE_VIDEO;
+ }
+
+ if (changedMask == 0) {
+ // If nothing changed as far as the audio/video decoders
+ // are concerned we can proceed.
+ onChangeConfiguration3(msg);
+ return;
+ }
+
+ // Something changed, inform the player which will shutdown the
+ // corresponding decoders and will post the reply once that's done.
+ // Handling the reply will continue executing below in
+ // onChangeConfiguration3.
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatStreamsChanged);
+ notify->setInt32("changedMask", changedMask);
+
+ msg->setWhat(kWhatChangeConfiguration3);
+ msg->setTarget(id());
+
+ notify->setMessage("reply", msg);
+ notify->post();
+}
+
+void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
+ // All remaining fetchers are still suspended, the player has shutdown
+ // any decoders that needed it.
+
+ uint32_t streamMask;
+ CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
+
+ AString audioURI, videoURI, subtitleURI;
+ if (streamMask & STREAMTYPE_AUDIO) {
+ CHECK(msg->findString("audioURI", &audioURI));
+ }
+ if (streamMask & STREAMTYPE_VIDEO) {
+ CHECK(msg->findString("videoURI", &videoURI));
+ }
+ if (streamMask & STREAMTYPE_SUBTITLES) {
+ CHECK(msg->findString("subtitleURI", &subtitleURI));
+ }
+
+ int64_t timeUs;
+ CHECK(msg->findInt64("timeUs", &timeUs));
+
+ if (timeUs < 0ll) {
+ timeUs = mLastDequeuedTimeUs;
+ }
+
+ mStreamMask = streamMask;
+ mAudioURI = audioURI;
+ mVideoURI = videoURI;
+ mSubtitleURI = subtitleURI;
+
+ // Resume all existing fetchers and assign them packet sources.
+ for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ const AString &uri = mFetcherInfos.keyAt(i);
+
+ uint32_t resumeMask = 0;
+
+ sp<AnotherPacketSource> audioSource;
+ if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) {
+ audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
+ resumeMask |= STREAMTYPE_AUDIO;
+ }
+
+ sp<AnotherPacketSource> videoSource;
+ if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) {
+ videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
+ resumeMask |= STREAMTYPE_VIDEO;
+ }
+
+ sp<AnotherPacketSource> subtitleSource;
+ if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) {
+ subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES);
+ resumeMask |= STREAMTYPE_SUBTITLES;
+ }
+
+ CHECK_NE(resumeMask, 0u);
+
+ ALOGV("resuming fetchers for mask 0x%08x", resumeMask);
+
+ streamMask &= ~resumeMask;
+
+ mFetcherInfos.valueAt(i).mFetcher->startAsync(
+ audioSource, videoSource, subtitleSource);
+ }
+
+ // streamMask now only contains the types that need a new fetcher created.
+
+ if (streamMask != 0) {
+ ALOGV("creating new fetchers for mask 0x%08x", streamMask);
+ }
+
+ while (streamMask != 0) {
+ StreamType streamType = (StreamType)(streamMask & ~(streamMask - 1));
+
+ AString uri;
+ switch (streamType) {
+ case STREAMTYPE_AUDIO:
+ uri = audioURI;
+ break;
+ case STREAMTYPE_VIDEO:
+ uri = videoURI;
+ break;
+ case STREAMTYPE_SUBTITLES:
+ uri = subtitleURI;
+ break;
+ default:
+ TRESPASS();
+ }
+
+ sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str());
+ CHECK(fetcher != NULL);
+
+ sp<AnotherPacketSource> audioSource;
+ if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) {
+ audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
+ audioSource->clear();
+
+ streamMask &= ~STREAMTYPE_AUDIO;
+ }
+
+ sp<AnotherPacketSource> videoSource;
+ if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) {
+ videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
+ videoSource->clear();
+
+ streamMask &= ~STREAMTYPE_VIDEO;
+ }
+
+ sp<AnotherPacketSource> subtitleSource;
+ if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) {
+ subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES);
+ subtitleSource->clear();
+
+ streamMask &= ~STREAMTYPE_SUBTITLES;
+ }
+
+ fetcher->startAsync(audioSource, videoSource, subtitleSource, timeUs);
+ }
+
+ // All fetchers have now been started, the configuration change
+ // has completed.
+
+ scheduleCheckBandwidthEvent();
+
+ ALOGV("XXX configuration change completed.");
+
+ mReconfigurationInProgress = false;
+
+ if (mDisconnectReplyID != 0) {
+ finishDisconnect();
+ }
+}
+
+void LiveSession::scheduleCheckBandwidthEvent() {
+ sp<AMessage> msg = new AMessage(kWhatCheckBandwidth, id());
+ msg->setInt32("generation", mCheckBandwidthGeneration);
+ msg->post(10000000ll);
+}
+
+void LiveSession::cancelCheckBandwidthEvent() {
+ ++mCheckBandwidthGeneration;
+}
+
+void LiveSession::onCheckBandwidth() {
+ if (mReconfigurationInProgress) {
+ scheduleCheckBandwidthEvent();
+ return;
+ }
+
+ size_t bandwidthIndex = getBandwidthIndex();
+ if (mPrevBandwidthIndex < 0
+ || bandwidthIndex != (size_t)mPrevBandwidthIndex) {
+ changeConfiguration(-1ll /* timeUs */, bandwidthIndex);
+ }
+
+ // Handling the kWhatCheckBandwidth even here does _not_ automatically
+ // schedule another one on return, only an explicit call to
+ // scheduleCheckBandwidthEvent will do that.
+ // This ensures that only one configuration change is ongoing at any
+ // one time, once that completes it'll schedule another check bandwidth
+ // event.
+}
+
+void LiveSession::postPrepared(status_t err) {
+ CHECK(mInPreparationPhase);
+
+ sp<AMessage> notify = mNotify->dup();
+ if (err == OK || err == ERROR_END_OF_STREAM) {
+ notify->setInt32("what", kWhatPrepared);
+ } else {
+ notify->setInt32("what", kWhatPreparationFailed);
+ notify->setInt32("err", err);
+ }
+
+ notify->post();
+
+ mInPreparationPhase = false;
}
} // namespace android
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
new file mode 100644
index 0000000..b134725
--- /dev/null
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -0,0 +1,172 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef LIVE_SESSION_H_
+
+#define LIVE_SESSION_H_
+
+#include <media/stagefright/foundation/AHandler.h>
+
+#include <utils/String8.h>
+
+namespace android {
+
+struct ABuffer;
+struct AnotherPacketSource;
+struct DataSource;
+struct HTTPBase;
+struct LiveDataSource;
+struct M3UParser;
+struct PlaylistFetcher;
+
+struct LiveSession : public AHandler {
+ enum Flags {
+ // Don't log any URLs.
+ kFlagIncognito = 1,
+ };
+ LiveSession(
+ const sp<AMessage> ¬ify,
+ uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
+
+ enum StreamType {
+ STREAMTYPE_AUDIO = 1,
+ STREAMTYPE_VIDEO = 2,
+ STREAMTYPE_SUBTITLES = 4,
+ };
+ status_t dequeueAccessUnit(StreamType stream, sp<ABuffer> *accessUnit);
+
+ status_t getStreamFormat(StreamType stream, sp<AMessage> *format);
+
+ void connectAsync(
+ const char *url,
+ const KeyedVector<String8, String8> *headers = NULL);
+
+ status_t disconnect();
+
+ // Blocks until seek is complete.
+ status_t seekTo(int64_t timeUs);
+
+ status_t getDuration(int64_t *durationUs) const;
+
+ bool isSeekable() const;
+ bool hasDynamicDuration() const;
+
+ enum {
+ kWhatStreamsChanged,
+ kWhatError,
+ kWhatPrepared,
+ kWhatPreparationFailed,
+ };
+
+protected:
+ virtual ~LiveSession();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ friend struct PlaylistFetcher;
+
+ enum {
+ kWhatConnect = 'conn',
+ kWhatDisconnect = 'disc',
+ kWhatSeek = 'seek',
+ kWhatFetcherNotify = 'notf',
+ kWhatCheckBandwidth = 'bndw',
+ kWhatChangeConfiguration2 = 'chC2',
+ kWhatChangeConfiguration3 = 'chC3',
+ kWhatFinishDisconnect2 = 'fin2',
+ };
+
+ struct BandwidthItem {
+ size_t mPlaylistIndex;
+ unsigned long mBandwidth;
+ };
+
+ struct FetcherInfo {
+ sp<PlaylistFetcher> mFetcher;
+ int64_t mDurationUs;
+ bool mIsPrepared;
+ };
+
+ sp<AMessage> mNotify;
+ uint32_t mFlags;
+ bool mUIDValid;
+ uid_t mUID;
+
+ bool mInPreparationPhase;
+
+ sp<HTTPBase> mHTTPDataSource;
+ KeyedVector<String8, String8> mExtraHeaders;
+
+ AString mMasterURL;
+
+ Vector<BandwidthItem> mBandwidthItems;
+ ssize_t mPrevBandwidthIndex;
+
+ sp<M3UParser> mPlaylist;
+
+ KeyedVector<AString, FetcherInfo> mFetcherInfos;
+ AString mAudioURI, mVideoURI, mSubtitleURI;
+ uint32_t mStreamMask;
+
+ KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources;
+
+ int32_t mCheckBandwidthGeneration;
+
+ size_t mContinuationCounter;
+ sp<AMessage> mContinuation;
+
+ int64_t mLastDequeuedTimeUs;
+
+ bool mReconfigurationInProgress;
+ uint32_t mDisconnectReplyID;
+
+ sp<PlaylistFetcher> addFetcher(const char *uri);
+
+ void onConnect(const sp<AMessage> &msg);
+ status_t onSeek(const sp<AMessage> &msg);
+ void onFinishDisconnect2();
+
+ status_t fetchFile(
+ const char *url, sp<ABuffer> *out,
+ int64_t range_offset = 0, int64_t range_length = -1);
+
+ sp<M3UParser> fetchPlaylist(
+ const char *url, uint8_t *curPlaylistHash, bool *unchanged);
+
+ size_t getBandwidthIndex();
+
+ static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *);
+
+ void changeConfiguration(int64_t timeUs, size_t bandwidthIndex);
+ void onChangeConfiguration2(const sp<AMessage> &msg);
+ void onChangeConfiguration3(const sp<AMessage> &msg);
+
+ void scheduleCheckBandwidthEvent();
+ void cancelCheckBandwidthEvent();
+
+ void onCheckBandwidth();
+
+ void finishDisconnect();
+
+ void postPrepared(status_t err);
+
+ DISALLOW_EVIL_CONSTRUCTORS(LiveSession);
+};
+
+} // namespace android
+
+#endif // LIVE_SESSION_H_
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 68bbca2..be66252 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -18,14 +18,153 @@
#define LOG_TAG "M3UParser"
#include <utils/Log.h>
-#include "include/M3UParser.h"
+#include "M3UParser.h"
+#include <cutils/properties.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MediaErrors.h>
namespace android {
+struct M3UParser::MediaGroup : public RefBase {
+ enum Type {
+ TYPE_AUDIO,
+ TYPE_VIDEO,
+ TYPE_SUBS,
+ };
+
+ enum FlagBits {
+ FLAG_AUTOSELECT = 1,
+ FLAG_DEFAULT = 2,
+ FLAG_FORCED = 4,
+ FLAG_HAS_LANGUAGE = 8,
+ FLAG_HAS_URI = 16,
+ };
+
+ MediaGroup(Type type);
+
+ Type type() const;
+
+ status_t addMedia(
+ const char *name,
+ const char *uri,
+ const char *language,
+ uint32_t flags);
+
+ bool getActiveURI(AString *uri) const;
+
+ void pickRandomMediaItems();
+
+protected:
+ virtual ~MediaGroup();
+
+private:
+ struct Media {
+ AString mName;
+ AString mURI;
+ AString mLanguage;
+ uint32_t mFlags;
+ };
+
+ Type mType;
+ Vector<Media> mMediaItems;
+
+ ssize_t mSelectedIndex;
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaGroup);
+};
+
+M3UParser::MediaGroup::MediaGroup(Type type)
+ : mType(type),
+ mSelectedIndex(-1) {
+}
+
+M3UParser::MediaGroup::~MediaGroup() {
+}
+
+M3UParser::MediaGroup::Type M3UParser::MediaGroup::type() const {
+ return mType;
+}
+
+status_t M3UParser::MediaGroup::addMedia(
+ const char *name,
+ const char *uri,
+ const char *language,
+ uint32_t flags) {
+ mMediaItems.push();
+ Media &item = mMediaItems.editItemAt(mMediaItems.size() - 1);
+
+ item.mName = name;
+
+ if (uri) {
+ item.mURI = uri;
+ }
+
+ if (language) {
+ item.mLanguage = language;
+ }
+
+ item.mFlags = flags;
+
+ return OK;
+}
+
+void M3UParser::MediaGroup::pickRandomMediaItems() {
+#if 1
+ switch (mType) {
+ case TYPE_AUDIO:
+ {
+ char value[PROPERTY_VALUE_MAX];
+ if (property_get("media.httplive.audio-index", value, NULL)) {
+ char *end;
+ mSelectedIndex = strtoul(value, &end, 10);
+ CHECK(end > value && *end == '\0');
+
+ if (mSelectedIndex >= mMediaItems.size()) {
+ mSelectedIndex = mMediaItems.size() - 1;
+ }
+ } else {
+ mSelectedIndex = 0;
+ }
+ break;
+ }
+
+ case TYPE_VIDEO:
+ {
+ mSelectedIndex = 0;
+ break;
+ }
+
+ case TYPE_SUBS:
+ {
+ mSelectedIndex = -1;
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+#else
+ mSelectedIndex = (rand() * mMediaItems.size()) / RAND_MAX;
+#endif
+}
+
+bool M3UParser::MediaGroup::getActiveURI(AString *uri) const {
+ for (size_t i = 0; i < mMediaItems.size(); ++i) {
+ if (mSelectedIndex >= 0 && i == (size_t)mSelectedIndex) {
+ const Media &item = mMediaItems.itemAt(i);
+
+ *uri = item.mURI;
+ return true;
+ }
+ }
+
+ return false;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
M3UParser::M3UParser(
const char *baseURI, const void *data, size_t size)
: mInitCheck(NO_INIT),
@@ -92,6 +231,58 @@
return true;
}
+void M3UParser::pickRandomMediaItems() {
+ for (size_t i = 0; i < mMediaGroups.size(); ++i) {
+ mMediaGroups.valueAt(i)->pickRandomMediaItems();
+ }
+}
+
+bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const {
+ if (!mIsVariantPlaylist) {
+ *uri = mBaseURI;
+
+ // Assume media without any more specific attribute contains
+ // audio and video, but no subtitles.
+ return !strcmp("audio", key) || !strcmp("video", key);
+ }
+
+ CHECK_LT(index, mItems.size());
+
+ sp<AMessage> meta = mItems.itemAt(index).mMeta;
+
+ AString groupID;
+ if (!meta->findString(key, &groupID)) {
+ *uri = mItems.itemAt(index).mURI;
+
+ // Assume media without any more specific attribute contains
+ // audio and video, but no subtitles.
+ return !strcmp("audio", key) || !strcmp("video", key);
+ }
+
+ sp<MediaGroup> group = mMediaGroups.valueFor(groupID);
+ if (!group->getActiveURI(uri)) {
+ return false;
+ }
+
+ if ((*uri).empty()) {
+ *uri = mItems.itemAt(index).mURI;
+ }
+
+ return true;
+}
+
+bool M3UParser::getAudioURI(size_t index, AString *uri) const {
+ return getTypeURI(index, "audio", uri);
+}
+
+bool M3UParser::getVideoURI(size_t index, AString *uri) const {
+ return getTypeURI(index, "video", uri);
+}
+
+bool M3UParser::getSubtitleURI(size_t index, AString *uri) const {
+ return getTypeURI(index, "subtitles", uri);
+}
+
static bool MakeURL(const char *baseURL, const char *url, AString *out) {
out->clear();
@@ -241,6 +432,8 @@
segmentRangeOffset = offset + length;
}
+ } else if (line.startsWith("#EXT-X-MEDIA")) {
+ err = parseMedia(line);
}
if (err != OK) {
@@ -322,9 +515,31 @@
return OK;
}
-// static
+// Find the next occurence of the character "what" at or after "offset",
+// but ignore occurences between quotation marks.
+// Return the index of the occurrence or -1 if not found.
+static ssize_t FindNextUnquoted(
+ const AString &line, char what, size_t offset) {
+ CHECK_NE((int)what, (int)'"');
+
+ bool quoted = false;
+ while (offset < line.size()) {
+ char c = line.c_str()[offset];
+
+ if (c == '"') {
+ quoted = !quoted;
+ } else if (c == what && !quoted) {
+ return offset;
+ }
+
+ ++offset;
+ }
+
+ return -1;
+}
+
status_t M3UParser::parseStreamInf(
- const AString &line, sp<AMessage> *meta) {
+ const AString &line, sp<AMessage> *meta) const {
ssize_t colonPos = line.find(":");
if (colonPos < 0) {
@@ -334,7 +549,7 @@
size_t offset = colonPos + 1;
while (offset < line.size()) {
- ssize_t end = line.find(",", offset);
+ ssize_t end = FindNextUnquoted(line, ',', offset);
if (end < 0) {
end = line.size();
}
@@ -371,35 +586,37 @@
*meta = new AMessage;
}
(*meta)->setInt32("bandwidth", x);
+ } else if (!strcasecmp("audio", key.c_str())
+ || !strcasecmp("video", key.c_str())
+ || !strcasecmp("subtitles", key.c_str())) {
+ if (val.size() < 2
+ || val.c_str()[0] != '"'
+ || val.c_str()[val.size() - 1] != '"') {
+ ALOGE("Expected quoted string for %s attribute, "
+ "got '%s' instead.",
+ key.c_str(), val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ AString groupID(val, 1, val.size() - 2);
+ ssize_t groupIndex = mMediaGroups.indexOfKey(groupID);
+
+ if (groupIndex < 0) {
+ ALOGE("Undefined media group '%s' referenced in stream info.",
+ groupID.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ key.tolower();
+ (*meta)->setString(key.c_str(), groupID.c_str());
}
}
return OK;
}
-// Find the next occurence of the character "what" at or after "offset",
-// but ignore occurences between quotation marks.
-// Return the index of the occurrence or -1 if not found.
-static ssize_t FindNextUnquoted(
- const AString &line, char what, size_t offset) {
- CHECK_NE((int)what, (int)'"');
-
- bool quoted = false;
- while (offset < line.size()) {
- char c = line.c_str()[offset];
-
- if (c == '"') {
- quoted = !quoted;
- } else if (c == what && !quoted) {
- return offset;
- }
-
- ++offset;
- }
-
- return -1;
-}
-
// static
status_t M3UParser::parseCipherInfo(
const AString &line, sp<AMessage> *meta, const AString &baseURI) {
@@ -515,6 +732,234 @@
return OK;
}
+status_t M3UParser::parseMedia(const AString &line) {
+ ssize_t colonPos = line.find(":");
+
+ if (colonPos < 0) {
+ return ERROR_MALFORMED;
+ }
+
+ bool haveGroupType = false;
+ MediaGroup::Type groupType = MediaGroup::TYPE_AUDIO;
+
+ bool haveGroupID = false;
+ AString groupID;
+
+ bool haveGroupLanguage = false;
+ AString groupLanguage;
+
+ bool haveGroupName = false;
+ AString groupName;
+
+ bool haveGroupAutoselect = false;
+ bool groupAutoselect = false;
+
+ bool haveGroupDefault = false;
+ bool groupDefault = false;
+
+ bool haveGroupForced = false;
+ bool groupForced = false;
+
+ bool haveGroupURI = false;
+ AString groupURI;
+
+ size_t offset = colonPos + 1;
+
+ while (offset < line.size()) {
+ ssize_t end = FindNextUnquoted(line, ',', offset);
+ if (end < 0) {
+ end = line.size();
+ }
+
+ AString attr(line, offset, end - offset);
+ attr.trim();
+
+ offset = end + 1;
+
+ ssize_t equalPos = attr.find("=");
+ if (equalPos < 0) {
+ continue;
+ }
+
+ AString key(attr, 0, equalPos);
+ key.trim();
+
+ AString val(attr, equalPos + 1, attr.size() - equalPos - 1);
+ val.trim();
+
+ ALOGV("key=%s value=%s", key.c_str(), val.c_str());
+
+ if (!strcasecmp("type", key.c_str())) {
+ if (!strcasecmp("subtitles", val.c_str())) {
+ groupType = MediaGroup::TYPE_SUBS;
+ } else if (!strcasecmp("audio", val.c_str())) {
+ groupType = MediaGroup::TYPE_AUDIO;
+ } else if (!strcasecmp("video", val.c_str())) {
+ groupType = MediaGroup::TYPE_VIDEO;
+ } else {
+ ALOGE("Invalid media group type '%s'", val.c_str());
+ return ERROR_MALFORMED;
+ }
+
+ haveGroupType = true;
+ } else if (!strcasecmp("group-id", key.c_str())) {
+ if (val.size() < 2
+ || val.c_str()[0] != '"'
+ || val.c_str()[val.size() - 1] != '"') {
+ ALOGE("Expected quoted string for GROUP-ID, got '%s' instead.",
+ val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ groupID.setTo(val, 1, val.size() - 2);
+ haveGroupID = true;
+ } else if (!strcasecmp("language", key.c_str())) {
+ if (val.size() < 2
+ || val.c_str()[0] != '"'
+ || val.c_str()[val.size() - 1] != '"') {
+ ALOGE("Expected quoted string for LANGUAGE, got '%s' instead.",
+ val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ groupLanguage.setTo(val, 1, val.size() - 2);
+ haveGroupLanguage = true;
+ } else if (!strcasecmp("name", key.c_str())) {
+ if (val.size() < 2
+ || val.c_str()[0] != '"'
+ || val.c_str()[val.size() - 1] != '"') {
+ ALOGE("Expected quoted string for NAME, got '%s' instead.",
+ val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ groupName.setTo(val, 1, val.size() - 2);
+ haveGroupName = true;
+ } else if (!strcasecmp("autoselect", key.c_str())) {
+ groupAutoselect = false;
+ if (!strcasecmp("YES", val.c_str())) {
+ groupAutoselect = true;
+ } else if (!strcasecmp("NO", val.c_str())) {
+ groupAutoselect = false;
+ } else {
+ ALOGE("Expected YES or NO for AUTOSELECT attribute, "
+ "got '%s' instead.",
+ val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ haveGroupAutoselect = true;
+ } else if (!strcasecmp("default", key.c_str())) {
+ groupDefault = false;
+ if (!strcasecmp("YES", val.c_str())) {
+ groupDefault = true;
+ } else if (!strcasecmp("NO", val.c_str())) {
+ groupDefault = false;
+ } else {
+ ALOGE("Expected YES or NO for DEFAULT attribute, "
+ "got '%s' instead.",
+ val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ haveGroupDefault = true;
+ } else if (!strcasecmp("forced", key.c_str())) {
+ groupForced = false;
+ if (!strcasecmp("YES", val.c_str())) {
+ groupForced = true;
+ } else if (!strcasecmp("NO", val.c_str())) {
+ groupForced = false;
+ } else {
+ ALOGE("Expected YES or NO for FORCED attribute, "
+ "got '%s' instead.",
+ val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ haveGroupForced = true;
+ } else if (!strcasecmp("uri", key.c_str())) {
+ if (val.size() < 2
+ || val.c_str()[0] != '"'
+ || val.c_str()[val.size() - 1] != '"') {
+ ALOGE("Expected quoted string for URI, got '%s' instead.",
+ val.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ AString tmp(val, 1, val.size() - 2);
+
+ if (!MakeURL(mBaseURI.c_str(), tmp.c_str(), &groupURI)) {
+ ALOGI("Failed to make absolute URI from '%s'.", tmp.c_str());
+ }
+
+ haveGroupURI = true;
+ }
+ }
+
+ if (!haveGroupType || !haveGroupID || !haveGroupName) {
+ ALOGE("Incomplete EXT-X-MEDIA element.");
+ return ERROR_MALFORMED;
+ }
+
+ uint32_t flags = 0;
+ if (haveGroupAutoselect && groupAutoselect) {
+ flags |= MediaGroup::FLAG_AUTOSELECT;
+ }
+ if (haveGroupDefault && groupDefault) {
+ flags |= MediaGroup::FLAG_DEFAULT;
+ }
+ if (haveGroupForced) {
+ if (groupType != MediaGroup::TYPE_SUBS) {
+ ALOGE("The FORCED attribute MUST not be present on anything "
+ "but SUBS media.");
+
+ return ERROR_MALFORMED;
+ }
+
+ if (groupForced) {
+ flags |= MediaGroup::FLAG_FORCED;
+ }
+ }
+ if (haveGroupLanguage) {
+ flags |= MediaGroup::FLAG_HAS_LANGUAGE;
+ }
+ if (haveGroupURI) {
+ flags |= MediaGroup::FLAG_HAS_URI;
+ }
+
+ ssize_t groupIndex = mMediaGroups.indexOfKey(groupID);
+ sp<MediaGroup> group;
+
+ if (groupIndex < 0) {
+ group = new MediaGroup(groupType);
+ mMediaGroups.add(groupID, group);
+ } else {
+ group = mMediaGroups.valueAt(groupIndex);
+
+ if (group->type() != groupType) {
+ ALOGE("Attempt to put media item under group of different type "
+ "(groupType = %d, item type = %d",
+ group->type(),
+ groupType);
+
+ return ERROR_MALFORMED;
+ }
+ }
+
+ return group->addMedia(
+ groupName.c_str(),
+ haveGroupURI ? groupURI.c_str() : NULL,
+ haveGroupLanguage ? groupLanguage.c_str() : NULL,
+ flags);
+}
+
// static
status_t M3UParser::ParseInt32(const char *s, int32_t *x) {
char *end;
diff --git a/media/libstagefright/include/M3UParser.h b/media/libstagefright/httplive/M3UParser.h
similarity index 80%
rename from media/libstagefright/include/M3UParser.h
rename to media/libstagefright/httplive/M3UParser.h
index 2d2f50f..abea286 100644
--- a/media/libstagefright/include/M3UParser.h
+++ b/media/libstagefright/httplive/M3UParser.h
@@ -40,10 +40,18 @@
size_t size();
bool itemAt(size_t index, AString *uri, sp<AMessage> *meta = NULL);
+ void pickRandomMediaItems();
+
+ bool getAudioURI(size_t index, AString *uri) const;
+ bool getVideoURI(size_t index, AString *uri) const;
+ bool getSubtitleURI(size_t index, AString *uri) const;
+
protected:
virtual ~M3UParser();
private:
+ struct MediaGroup;
+
struct Item {
AString mURI;
sp<AMessage> mMeta;
@@ -60,6 +68,9 @@
sp<AMessage> mMeta;
Vector<Item> mItems;
+ // Media groups keyed by group ID.
+ KeyedVector<AString, sp<MediaGroup> > mMediaGroups;
+
status_t parse(const void *data, size_t size);
static status_t parseMetaData(
@@ -68,8 +79,8 @@
static status_t parseMetaDataDuration(
const AString &line, sp<AMessage> *meta, const char *key);
- static status_t parseStreamInf(
- const AString &line, sp<AMessage> *meta);
+ status_t parseStreamInf(
+ const AString &line, sp<AMessage> *meta) const;
static status_t parseCipherInfo(
const AString &line, sp<AMessage> *meta, const AString &baseURI);
@@ -78,6 +89,10 @@
const AString &line, uint64_t curOffset,
uint64_t *length, uint64_t *offset);
+ status_t parseMedia(const AString &line);
+
+ bool getTypeURI(size_t index, const char *key, AString *uri) const;
+
static status_t ParseInt32(const char *s, int32_t *x);
static status_t ParseDouble(const char *s, double *x);
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
new file mode 100644
index 0000000..8ae70b7
--- /dev/null
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -0,0 +1,969 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "PlaylistFetcher"
+#include <utils/Log.h>
+
+#include "PlaylistFetcher.h"
+
+#include "LiveDataSource.h"
+#include "LiveSession.h"
+#include "M3UParser.h"
+
+#include "include/avc_utils.h"
+#include "include/HTTPBase.h"
+#include "include/ID3.h"
+#include "mpeg2ts/AnotherPacketSource.h"
+
+#include <media/IStreamSource.h>
+#include <media/stagefright/foundation/ABitReader.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/FileSource.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+#include <ctype.h>
+#include <openssl/aes.h>
+#include <openssl/md5.h>
+
+namespace android {
+
+// static
+const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
+
+PlaylistFetcher::PlaylistFetcher(
+ const sp<AMessage> ¬ify,
+ const sp<LiveSession> &session,
+ const char *uri)
+ : mNotify(notify),
+ mSession(session),
+ mURI(uri),
+ mStreamTypeMask(0),
+ mStartTimeUs(-1ll),
+ mLastPlaylistFetchTimeUs(-1ll),
+ mSeqNumber(-1),
+ mNumRetries(0),
+ mStartup(true),
+ mNextPTSTimeUs(-1ll),
+ mMonitorQueueGeneration(0),
+ mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY),
+ mFirstPTSValid(false),
+ mAbsoluteTimeAnchorUs(0ll) {
+ memset(mPlaylistHash, 0, sizeof(mPlaylistHash));
+}
+
+PlaylistFetcher::~PlaylistFetcher() {
+}
+
+int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const {
+ CHECK(mPlaylist != NULL);
+
+ int32_t firstSeqNumberInPlaylist;
+ if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
+ "media-sequence", &firstSeqNumberInPlaylist)) {
+ firstSeqNumberInPlaylist = 0;
+ }
+
+ int32_t lastSeqNumberInPlaylist =
+ firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
+
+ CHECK_GE(seqNumber, firstSeqNumberInPlaylist);
+ CHECK_LE(seqNumber, lastSeqNumberInPlaylist);
+
+ int64_t segmentStartUs = 0ll;
+ for (int32_t index = 0;
+ index < seqNumber - firstSeqNumberInPlaylist; ++index) {
+ sp<AMessage> itemMeta;
+ CHECK(mPlaylist->itemAt(
+ index, NULL /* uri */, &itemMeta));
+
+ int64_t itemDurationUs;
+ CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
+
+ segmentStartUs += itemDurationUs;
+ }
+
+ return segmentStartUs;
+}
+
+bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const {
+ if (mPlaylist == NULL) {
+ CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY);
+ return true;
+ }
+
+ int32_t targetDurationSecs;
+ CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+
+ int64_t targetDurationUs = targetDurationSecs * 1000000ll;
+
+ int64_t minPlaylistAgeUs;
+
+ switch (mRefreshState) {
+ case INITIAL_MINIMUM_RELOAD_DELAY:
+ {
+ size_t n = mPlaylist->size();
+ if (n > 0) {
+ sp<AMessage> itemMeta;
+ CHECK(mPlaylist->itemAt(n - 1, NULL /* uri */, &itemMeta));
+
+ int64_t itemDurationUs;
+ CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
+
+ minPlaylistAgeUs = itemDurationUs;
+ break;
+ }
+
+ // fall through
+ }
+
+ case FIRST_UNCHANGED_RELOAD_ATTEMPT:
+ {
+ minPlaylistAgeUs = targetDurationUs / 2;
+ break;
+ }
+
+ case SECOND_UNCHANGED_RELOAD_ATTEMPT:
+ {
+ minPlaylistAgeUs = (targetDurationUs * 3) / 2;
+ break;
+ }
+
+ case THIRD_UNCHANGED_RELOAD_ATTEMPT:
+ {
+ minPlaylistAgeUs = targetDurationUs * 3;
+ break;
+ }
+
+ default:
+ TRESPASS();
+ break;
+ }
+
+ return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs;
+}
+
+status_t PlaylistFetcher::decryptBuffer(
+ size_t playlistIndex, const sp<ABuffer> &buffer) {
+ sp<AMessage> itemMeta;
+ bool found = false;
+ AString method;
+
+ for (ssize_t i = playlistIndex; i >= 0; --i) {
+ AString uri;
+ CHECK(mPlaylist->itemAt(i, &uri, &itemMeta));
+
+ if (itemMeta->findString("cipher-method", &method)) {
+ found = true;
+ break;
+ }
+ }
+
+ if (!found) {
+ method = "NONE";
+ }
+
+ if (method == "NONE") {
+ return OK;
+ } else if (!(method == "AES-128")) {
+ ALOGE("Unsupported cipher method '%s'", method.c_str());
+ return ERROR_UNSUPPORTED;
+ }
+
+ AString keyURI;
+ if (!itemMeta->findString("cipher-uri", &keyURI)) {
+ ALOGE("Missing key uri");
+ return ERROR_MALFORMED;
+ }
+
+ ssize_t index = mAESKeyForURI.indexOfKey(keyURI);
+
+ sp<ABuffer> key;
+ if (index >= 0) {
+ key = mAESKeyForURI.valueAt(index);
+ } else {
+ status_t err = mSession->fetchFile(keyURI.c_str(), &key);
+
+ if (err != OK) {
+ ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str());
+ return ERROR_IO;
+ } else if (key->size() != 16) {
+ ALOGE("key file '%s' wasn't 16 bytes in size.", keyURI.c_str());
+ return ERROR_MALFORMED;
+ }
+
+ mAESKeyForURI.add(keyURI, key);
+ }
+
+ AES_KEY aes_key;
+ if (AES_set_decrypt_key(key->data(), 128, &aes_key) != 0) {
+ ALOGE("failed to set AES decryption key.");
+ return UNKNOWN_ERROR;
+ }
+
+ unsigned char aes_ivec[16];
+
+ AString iv;
+ if (itemMeta->findString("cipher-iv", &iv)) {
+ if ((!iv.startsWith("0x") && !iv.startsWith("0X"))
+ || iv.size() != 16 * 2 + 2) {
+ ALOGE("malformed cipher IV '%s'.", iv.c_str());
+ return ERROR_MALFORMED;
+ }
+
+ memset(aes_ivec, 0, sizeof(aes_ivec));
+ for (size_t i = 0; i < 16; ++i) {
+ char c1 = tolower(iv.c_str()[2 + 2 * i]);
+ char c2 = tolower(iv.c_str()[3 + 2 * i]);
+ if (!isxdigit(c1) || !isxdigit(c2)) {
+ ALOGE("malformed cipher IV '%s'.", iv.c_str());
+ return ERROR_MALFORMED;
+ }
+ uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10;
+ uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10;
+
+ aes_ivec[i] = nibble1 << 4 | nibble2;
+ }
+ } else {
+ memset(aes_ivec, 0, sizeof(aes_ivec));
+ aes_ivec[15] = mSeqNumber & 0xff;
+ aes_ivec[14] = (mSeqNumber >> 8) & 0xff;
+ aes_ivec[13] = (mSeqNumber >> 16) & 0xff;
+ aes_ivec[12] = (mSeqNumber >> 24) & 0xff;
+ }
+
+ AES_cbc_encrypt(
+ buffer->data(), buffer->data(), buffer->size(),
+ &aes_key, aes_ivec, AES_DECRYPT);
+
+ // hexdump(buffer->data(), buffer->size());
+
+ size_t n = buffer->size();
+ CHECK_GT(n, 0u);
+
+ size_t pad = buffer->data()[n - 1];
+
+ CHECK_GT(pad, 0u);
+ CHECK_LE(pad, 16u);
+ CHECK_GE((size_t)n, pad);
+ for (size_t i = 0; i < pad; ++i) {
+ CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad);
+ }
+
+ n -= pad;
+
+ buffer->setRange(buffer->offset(), n);
+
+ return OK;
+}
+
+void PlaylistFetcher::postMonitorQueue(int64_t delayUs) {
+ sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id());
+ msg->setInt32("generation", mMonitorQueueGeneration);
+ msg->post(delayUs);
+}
+
+void PlaylistFetcher::cancelMonitorQueue() {
+ ++mMonitorQueueGeneration;
+}
+
+void PlaylistFetcher::startAsync(
+ const sp<AnotherPacketSource> &audioSource,
+ const sp<AnotherPacketSource> &videoSource,
+ const sp<AnotherPacketSource> &subtitleSource,
+ int64_t startTimeUs) {
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+
+ uint32_t streamTypeMask = 0ul;
+
+ if (audioSource != NULL) {
+ msg->setPointer("audioSource", audioSource.get());
+ streamTypeMask |= LiveSession::STREAMTYPE_AUDIO;
+ }
+
+ if (videoSource != NULL) {
+ msg->setPointer("videoSource", videoSource.get());
+ streamTypeMask |= LiveSession::STREAMTYPE_VIDEO;
+ }
+
+ if (subtitleSource != NULL) {
+ msg->setPointer("subtitleSource", subtitleSource.get());
+ streamTypeMask |= LiveSession::STREAMTYPE_SUBTITLES;
+ }
+
+ msg->setInt32("streamTypeMask", streamTypeMask);
+ msg->setInt64("startTimeUs", startTimeUs);
+ msg->post();
+}
+
+void PlaylistFetcher::pauseAsync() {
+ (new AMessage(kWhatPause, id()))->post();
+}
+
+void PlaylistFetcher::stopAsync() {
+ (new AMessage(kWhatStop, id()))->post();
+}
+
+void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatStart:
+ {
+ status_t err = onStart(msg);
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatStarted);
+ notify->setInt32("err", err);
+ notify->post();
+ break;
+ }
+
+ case kWhatPause:
+ {
+ onPause();
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatPaused);
+ notify->post();
+ break;
+ }
+
+ case kWhatStop:
+ {
+ onStop();
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatStopped);
+ notify->post();
+ break;
+ }
+
+ case kWhatMonitorQueue:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mMonitorQueueGeneration) {
+ // Stale event
+ break;
+ }
+
+ onMonitorQueue();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) {
+ mPacketSources.clear();
+
+ uint32_t streamTypeMask;
+ CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask));
+
+ int64_t startTimeUs;
+ CHECK(msg->findInt64("startTimeUs", &startTimeUs));
+
+ if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) {
+ void *ptr;
+ CHECK(msg->findPointer("audioSource", &ptr));
+
+ mPacketSources.add(
+ LiveSession::STREAMTYPE_AUDIO,
+ static_cast<AnotherPacketSource *>(ptr));
+ }
+
+ if (streamTypeMask & LiveSession::STREAMTYPE_VIDEO) {
+ void *ptr;
+ CHECK(msg->findPointer("videoSource", &ptr));
+
+ mPacketSources.add(
+ LiveSession::STREAMTYPE_VIDEO,
+ static_cast<AnotherPacketSource *>(ptr));
+ }
+
+ if (streamTypeMask & LiveSession::STREAMTYPE_SUBTITLES) {
+ void *ptr;
+ CHECK(msg->findPointer("subtitleSource", &ptr));
+
+ mPacketSources.add(
+ LiveSession::STREAMTYPE_SUBTITLES,
+ static_cast<AnotherPacketSource *>(ptr));
+ }
+
+ mStreamTypeMask = streamTypeMask;
+ mStartTimeUs = startTimeUs;
+
+ if (mStartTimeUs >= 0ll) {
+ mSeqNumber = -1;
+ mStartup = true;
+ }
+
+ postMonitorQueue();
+
+ return OK;
+}
+
+void PlaylistFetcher::onPause() {
+ cancelMonitorQueue();
+
+ mPacketSources.clear();
+ mStreamTypeMask = 0;
+}
+
+void PlaylistFetcher::onStop() {
+ cancelMonitorQueue();
+
+ for (size_t i = 0; i < mPacketSources.size(); ++i) {
+ mPacketSources.valueAt(i)->clear();
+ }
+
+ mPacketSources.clear();
+ mStreamTypeMask = 0;
+}
+
+void PlaylistFetcher::notifyError(status_t err) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatError);
+ notify->setInt32("err", err);
+ notify->post();
+}
+
+void PlaylistFetcher::queueDiscontinuity(
+ ATSParser::DiscontinuityType type, const sp<AMessage> &extra) {
+ for (size_t i = 0; i < mPacketSources.size(); ++i) {
+ mPacketSources.valueAt(i)->queueDiscontinuity(type, extra);
+ }
+}
+
+void PlaylistFetcher::onMonitorQueue() {
+ bool downloadMore = false;
+
+ status_t finalResult;
+ if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
+ sp<AnotherPacketSource> packetSource =
+ mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
+
+ downloadMore = packetSource->hasBufferAvailable(&finalResult);
+ } else {
+ bool first = true;
+ int64_t minBufferedDurationUs = 0ll;
+
+ for (size_t i = 0; i < mPacketSources.size(); ++i) {
+ if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
+ continue;
+ }
+
+ int64_t bufferedDurationUs =
+ mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
+
+ if (first || bufferedDurationUs < minBufferedDurationUs) {
+ minBufferedDurationUs = bufferedDurationUs;
+ first = false;
+ }
+ }
+
+ downloadMore =
+ !first && (minBufferedDurationUs < kMinBufferedDurationUs);
+ }
+
+ if (finalResult == OK && downloadMore) {
+ onDownloadNext();
+ } else {
+ // Nothing to do yet, try again in a second.
+
+ sp<AMessage> msg = mNotify->dup();
+ msg->setInt32("what", kWhatTemporarilyDoneFetching);
+ msg->post();
+
+ postMonitorQueue(1000000ll);
+ }
+}
+
+void PlaylistFetcher::onDownloadNext() {
+ int64_t nowUs = ALooper::GetNowUs();
+
+ if (mLastPlaylistFetchTimeUs < 0ll
+ || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) {
+ bool unchanged;
+ sp<M3UParser> playlist = mSession->fetchPlaylist(
+ mURI.c_str(), mPlaylistHash, &unchanged);
+
+ if (playlist == NULL) {
+ if (unchanged) {
+ // We succeeded in fetching the playlist, but it was
+ // unchanged from the last time we tried.
+
+ if (mRefreshState != THIRD_UNCHANGED_RELOAD_ATTEMPT) {
+ mRefreshState = (RefreshState)(mRefreshState + 1);
+ }
+ } else {
+ ALOGE("failed to load playlist at url '%s'", mURI.c_str());
+ notifyError(ERROR_IO);
+ return;
+ }
+ } else {
+ mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY;
+ mPlaylist = playlist;
+
+ if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
+ updateDuration();
+ }
+ }
+
+ mLastPlaylistFetchTimeUs = ALooper::GetNowUs();
+ }
+
+ int32_t firstSeqNumberInPlaylist;
+ if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
+ "media-sequence", &firstSeqNumberInPlaylist)) {
+ firstSeqNumberInPlaylist = 0;
+ }
+
+ bool seekDiscontinuity = false;
+ bool explicitDiscontinuity = false;
+
+ const int32_t lastSeqNumberInPlaylist =
+ firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
+
+ if (mSeqNumber < 0) {
+ CHECK_GE(mStartTimeUs, 0ll);
+
+ if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
+ mSeqNumber = getSeqNumberForTime(mStartTimeUs);
+ } else {
+ // If this is a live session, start 3 segments from the end.
+ mSeqNumber = lastSeqNumberInPlaylist - 3;
+ if (mSeqNumber < firstSeqNumberInPlaylist) {
+ mSeqNumber = firstSeqNumberInPlaylist;
+ }
+ }
+
+ mStartTimeUs = -1ll;
+ }
+
+ if (mSeqNumber < firstSeqNumberInPlaylist
+ || mSeqNumber > lastSeqNumberInPlaylist) {
+ if (!mPlaylist->isComplete() && mNumRetries < kMaxNumRetries) {
+ ++mNumRetries;
+
+ if (mSeqNumber > lastSeqNumberInPlaylist) {
+ mLastPlaylistFetchTimeUs = -1;
+ postMonitorQueue(3000000ll);
+ return;
+ }
+
+ // we've missed the boat, let's start from the lowest sequence
+ // number available and signal a discontinuity.
+
+ ALOGI("We've missed the boat, restarting playback.");
+ mSeqNumber = lastSeqNumberInPlaylist;
+ explicitDiscontinuity = true;
+
+ // fall through
+ } else {
+ ALOGE("Cannot find sequence number %d in playlist "
+ "(contains %d - %d)",
+ mSeqNumber, firstSeqNumberInPlaylist,
+ firstSeqNumberInPlaylist + mPlaylist->size() - 1);
+
+ notifyError(ERROR_END_OF_STREAM);
+ return;
+ }
+ }
+
+ mNumRetries = 0;
+
+ AString uri;
+ sp<AMessage> itemMeta;
+ CHECK(mPlaylist->itemAt(
+ mSeqNumber - firstSeqNumberInPlaylist,
+ &uri,
+ &itemMeta));
+
+ int32_t val;
+ if (itemMeta->findInt32("discontinuity", &val) && val != 0) {
+ explicitDiscontinuity = true;
+ }
+
+ int64_t range_offset, range_length;
+ if (!itemMeta->findInt64("range-offset", &range_offset)
+ || !itemMeta->findInt64("range-length", &range_length)) {
+ range_offset = 0;
+ range_length = -1;
+ }
+
+ ALOGV("fetching segment %d from (%d .. %d)",
+ mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist);
+
+ ALOGV("fetching '%s'", uri.c_str());
+
+ sp<ABuffer> buffer;
+ status_t err = mSession->fetchFile(
+ uri.c_str(), &buffer, range_offset, range_length);
+
+ if (err != OK) {
+ ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());
+ notifyError(err);
+ return;
+ }
+
+ CHECK(buffer != NULL);
+
+ err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer);
+
+ if (err != OK) {
+ ALOGE("decryptBuffer failed w/ error %d", err);
+
+ notifyError(err);
+ return;
+ }
+
+ if (mStartup || seekDiscontinuity || explicitDiscontinuity) {
+ // Signal discontinuity.
+
+ if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
+ // If this was a live event this made no sense since
+ // we don't have access to all the segment before the current
+ // one.
+ mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber);
+ }
+
+ if (seekDiscontinuity || explicitDiscontinuity) {
+ ALOGI("queueing discontinuity (seek=%d, explicit=%d)",
+ seekDiscontinuity, explicitDiscontinuity);
+
+ queueDiscontinuity(
+ explicitDiscontinuity
+ ? ATSParser::DISCONTINUITY_FORMATCHANGE
+ : ATSParser::DISCONTINUITY_SEEK,
+ NULL /* extra */);
+ }
+ }
+
+ err = extractAndQueueAccessUnits(buffer);
+
+ if (err != OK) {
+ notifyError(err);
+ return;
+ }
+
+ ++mSeqNumber;
+
+ postMonitorQueue();
+
+ mStartup = false;
+}
+
+int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const {
+ int32_t firstSeqNumberInPlaylist;
+ if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
+ "media-sequence", &firstSeqNumberInPlaylist)) {
+ firstSeqNumberInPlaylist = 0;
+ }
+
+ size_t index = 0;
+ int64_t segmentStartUs = 0;
+ while (index < mPlaylist->size()) {
+ sp<AMessage> itemMeta;
+ CHECK(mPlaylist->itemAt(
+ index, NULL /* uri */, &itemMeta));
+
+ int64_t itemDurationUs;
+ CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
+
+ if (timeUs < segmentStartUs + itemDurationUs) {
+ break;
+ }
+
+ segmentStartUs += itemDurationUs;
+ ++index;
+ }
+
+ if (index >= mPlaylist->size()) {
+ index = mPlaylist->size() - 1;
+ }
+
+ return firstSeqNumberInPlaylist + index;
+}
+
+status_t PlaylistFetcher::extractAndQueueAccessUnits(
+ const sp<ABuffer> &buffer) {
+ if (buffer->size() > 0 && buffer->data()[0] == 0x47) {
+ // Let's assume this is an MPEG2 transport stream.
+
+ if ((buffer->size() % 188) != 0) {
+ ALOGE("MPEG2 transport stream is not an even multiple of 188 "
+ "bytes in length.");
+ return ERROR_MALFORMED;
+ }
+
+ if (mTSParser == NULL) {
+ mTSParser = new ATSParser;
+ }
+
+ if (mNextPTSTimeUs >= 0ll) {
+ sp<AMessage> extra = new AMessage;
+ extra->setInt64(IStreamListener::kKeyMediaTimeUs, mNextPTSTimeUs);
+
+ mTSParser->signalDiscontinuity(
+ ATSParser::DISCONTINUITY_SEEK, extra);
+
+ mNextPTSTimeUs = -1ll;
+ }
+
+ size_t offset = 0;
+ while (offset < buffer->size()) {
+ status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188);
+
+ if (err != OK) {
+ return err;
+ }
+
+ offset += 188;
+ }
+
+ for (size_t i = mPacketSources.size(); i-- > 0;) {
+ sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
+
+ ATSParser::SourceType type;
+ switch (mPacketSources.keyAt(i)) {
+ case LiveSession::STREAMTYPE_VIDEO:
+ type = ATSParser::VIDEO;
+ break;
+
+ case LiveSession::STREAMTYPE_AUDIO:
+ type = ATSParser::AUDIO;
+ break;
+
+ case LiveSession::STREAMTYPE_SUBTITLES:
+ {
+ ALOGE("MPEG2 Transport streams do not contain subtitles.");
+ return ERROR_MALFORMED;
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+
+ sp<AnotherPacketSource> source =
+ static_cast<AnotherPacketSource *>(
+ mTSParser->getSource(type).get());
+
+ if (source == NULL) {
+ ALOGW("MPEG2 Transport stream does not contain %s data.",
+ type == ATSParser::VIDEO ? "video" : "audio");
+
+ mStreamTypeMask &= ~mPacketSources.keyAt(i);
+ mPacketSources.removeItemsAt(i);
+ continue;
+ }
+
+ sp<ABuffer> accessUnit;
+ status_t finalResult;
+ while (source->hasBufferAvailable(&finalResult)
+ && source->dequeueAccessUnit(&accessUnit) == OK) {
+ // Note that we do NOT dequeue any discontinuities.
+
+ packetSource->queueAccessUnit(accessUnit);
+ }
+
+ if (packetSource->getFormat() == NULL) {
+ packetSource->setFormat(source->getFormat());
+ }
+ }
+
+ return OK;
+ } else if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) {
+ if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) {
+ ALOGE("This stream only contains subtitles.");
+ return ERROR_MALFORMED;
+ }
+
+ const sp<AnotherPacketSource> packetSource =
+ mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
+
+ buffer->meta()->setInt64("timeUs", 0ll);
+
+ packetSource->queueAccessUnit(buffer);
+ return OK;
+ }
+
+ if (mNextPTSTimeUs >= 0ll) {
+ mFirstPTSValid = false;
+ mAbsoluteTimeAnchorUs = mNextPTSTimeUs;
+ mNextPTSTimeUs = -1ll;
+ }
+
+ // This better be an ISO 13818-7 (AAC) or ISO 13818-1 (MPEG) audio
+ // stream prefixed by an ID3 tag.
+
+ bool firstID3Tag = true;
+ uint64_t PTS = 0;
+
+ for (;;) {
+ // Make sure to skip all ID3 tags preceding the audio data.
+ // At least one must be present to provide the PTS timestamp.
+
+ ID3 id3(buffer->data(), buffer->size(), true /* ignoreV1 */);
+ if (!id3.isValid()) {
+ if (firstID3Tag) {
+ ALOGE("Unable to parse ID3 tag.");
+ return ERROR_MALFORMED;
+ } else {
+ break;
+ }
+ }
+
+ if (firstID3Tag) {
+ bool found = false;
+
+ ID3::Iterator it(id3, "PRIV");
+ while (!it.done()) {
+ size_t length;
+ const uint8_t *data = it.getData(&length);
+
+ static const char *kMatchName =
+ "com.apple.streaming.transportStreamTimestamp";
+ static const size_t kMatchNameLen = strlen(kMatchName);
+
+ if (length == kMatchNameLen + 1 + 8
+ && !strncmp((const char *)data, kMatchName, kMatchNameLen)) {
+ found = true;
+ PTS = U64_AT(&data[kMatchNameLen + 1]);
+ }
+
+ it.next();
+ }
+
+ if (!found) {
+ ALOGE("Unable to extract transportStreamTimestamp from ID3 tag.");
+ return ERROR_MALFORMED;
+ }
+ }
+
+ // skip the ID3 tag
+ buffer->setRange(
+ buffer->offset() + id3.rawSize(), buffer->size() - id3.rawSize());
+
+ firstID3Tag = false;
+ }
+
+ if (!mFirstPTSValid) {
+ mFirstPTSValid = true;
+ mFirstPTS = PTS;
+ }
+ PTS -= mFirstPTS;
+
+ int64_t timeUs = (PTS * 100ll) / 9ll + mAbsoluteTimeAnchorUs;
+
+ if (mStreamTypeMask != LiveSession::STREAMTYPE_AUDIO) {
+ ALOGW("This stream only contains audio data!");
+
+ mStreamTypeMask &= LiveSession::STREAMTYPE_AUDIO;
+
+ if (mStreamTypeMask == 0) {
+ return OK;
+ }
+ }
+
+ sp<AnotherPacketSource> packetSource =
+ mPacketSources.valueFor(LiveSession::STREAMTYPE_AUDIO);
+
+ if (packetSource->getFormat() == NULL && buffer->size() >= 7) {
+ ABitReader bits(buffer->data(), buffer->size());
+
+ // adts_fixed_header
+
+ CHECK_EQ(bits.getBits(12), 0xfffu);
+ bits.skipBits(3); // ID, layer
+ bool protection_absent = bits.getBits(1) != 0;
+
+ unsigned profile = bits.getBits(2);
+ CHECK_NE(profile, 3u);
+ unsigned sampling_freq_index = bits.getBits(4);
+ bits.getBits(1); // private_bit
+ unsigned channel_configuration = bits.getBits(3);
+ CHECK_NE(channel_configuration, 0u);
+ bits.skipBits(2); // original_copy, home
+
+ sp<MetaData> meta = MakeAACCodecSpecificData(
+ profile, sampling_freq_index, channel_configuration);
+
+ meta->setInt32(kKeyIsADTS, true);
+
+ packetSource->setFormat(meta);
+ }
+
+ int64_t numSamples = 0ll;
+ int32_t sampleRate;
+ CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate));
+
+ size_t offset = 0;
+ while (offset < buffer->size()) {
+ const uint8_t *adtsHeader = buffer->data() + offset;
+ CHECK_LT(offset + 5, buffer->size());
+
+ unsigned aac_frame_length =
+ ((adtsHeader[3] & 3) << 11)
+ | (adtsHeader[4] << 3)
+ | (adtsHeader[5] >> 5);
+
+ CHECK_LE(offset + aac_frame_length, buffer->size());
+
+ sp<ABuffer> unit = new ABuffer(aac_frame_length);
+ memcpy(unit->data(), adtsHeader, aac_frame_length);
+
+ int64_t unitTimeUs = timeUs + numSamples * 1000000ll / sampleRate;
+ unit->meta()->setInt64("timeUs", unitTimeUs);
+
+ // Each AAC frame encodes 1024 samples.
+ numSamples += 1024;
+
+ packetSource->queueAccessUnit(unit);
+
+ offset += aac_frame_length;
+ }
+
+ return OK;
+}
+
+void PlaylistFetcher::updateDuration() {
+ int64_t durationUs = 0ll;
+ for (size_t index = 0; index < mPlaylist->size(); ++index) {
+ sp<AMessage> itemMeta;
+ CHECK(mPlaylist->itemAt(
+ index, NULL /* uri */, &itemMeta));
+
+ int64_t itemDurationUs;
+ CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
+
+ durationUs += itemDurationUs;
+ }
+
+ sp<AMessage> msg = mNotify->dup();
+ msg->setInt32("what", kWhatDurationUpdate);
+ msg->setInt64("durationUs", durationUs);
+ msg->post();
+}
+
+} // namespace android
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
new file mode 100644
index 0000000..5a2b901
--- /dev/null
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef PLAYLIST_FETCHER_H_
+
+#define PLAYLIST_FETCHER_H_
+
+#include <media/stagefright/foundation/AHandler.h>
+
+#include "mpeg2ts/ATSParser.h"
+#include "LiveSession.h"
+
+namespace android {
+
+struct ABuffer;
+struct AnotherPacketSource;
+struct DataSource;
+struct HTTPBase;
+struct LiveDataSource;
+struct M3UParser;
+struct String8;
+
+struct PlaylistFetcher : public AHandler {
+ enum {
+ kWhatStarted,
+ kWhatPaused,
+ kWhatStopped,
+ kWhatError,
+ kWhatDurationUpdate,
+ kWhatTemporarilyDoneFetching,
+ kWhatPrepared,
+ kWhatPreparationFailed,
+ };
+
+ PlaylistFetcher(
+ const sp<AMessage> ¬ify,
+ const sp<LiveSession> &session,
+ const char *uri);
+
+ sp<DataSource> getDataSource();
+
+ void startAsync(
+ const sp<AnotherPacketSource> &audioSource,
+ const sp<AnotherPacketSource> &videoSource,
+ const sp<AnotherPacketSource> &subtitleSource,
+ int64_t startTimeUs = -1ll);
+
+ void pauseAsync();
+
+ void stopAsync();
+
+protected:
+ virtual ~PlaylistFetcher();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kMaxNumRetries = 5,
+ };
+
+ enum {
+ kWhatStart = 'strt',
+ kWhatPause = 'paus',
+ kWhatStop = 'stop',
+ kWhatMonitorQueue = 'moni',
+ };
+
+ static const int64_t kMinBufferedDurationUs;
+
+ sp<AMessage> mNotify;
+ sp<LiveSession> mSession;
+ AString mURI;
+
+ uint32_t mStreamTypeMask;
+ int64_t mStartTimeUs;
+
+ KeyedVector<LiveSession::StreamType, sp<AnotherPacketSource> >
+ mPacketSources;
+
+ KeyedVector<AString, sp<ABuffer> > mAESKeyForURI;
+
+ int64_t mLastPlaylistFetchTimeUs;
+ sp<M3UParser> mPlaylist;
+ int32_t mSeqNumber;
+ int32_t mNumRetries;
+ bool mStartup;
+ int64_t mNextPTSTimeUs;
+
+ int32_t mMonitorQueueGeneration;
+
+ enum RefreshState {
+ INITIAL_MINIMUM_RELOAD_DELAY,
+ FIRST_UNCHANGED_RELOAD_ATTEMPT,
+ SECOND_UNCHANGED_RELOAD_ATTEMPT,
+ THIRD_UNCHANGED_RELOAD_ATTEMPT
+ };
+ RefreshState mRefreshState;
+
+ uint8_t mPlaylistHash[16];
+
+ sp<ATSParser> mTSParser;
+
+ bool mFirstPTSValid;
+ uint64_t mFirstPTS;
+ int64_t mAbsoluteTimeAnchorUs;
+
+ status_t decryptBuffer(
+ size_t playlistIndex, const sp<ABuffer> &buffer);
+
+ void postMonitorQueue(int64_t delayUs = 0);
+ void cancelMonitorQueue();
+
+ bool timeToRefreshPlaylist(int64_t nowUs) const;
+
+ // Returns the media time in us of the segment specified by seqNumber.
+ // This is computed by summing the durations of all segments before it.
+ int64_t getSegmentStartTimeUs(int32_t seqNumber) const;
+
+ status_t onStart(const sp<AMessage> &msg);
+ void onPause();
+ void onStop();
+ void onMonitorQueue();
+ void onDownloadNext();
+
+ status_t extractAndQueueAccessUnits(const sp<ABuffer> &buffer);
+
+ void notifyError(status_t err);
+
+ void queueDiscontinuity(
+ ATSParser::DiscontinuityType type, const sp<AMessage> &extra);
+
+ int32_t getSeqNumberForTime(int64_t timeUs) const;
+
+ void updateDuration();
+
+ DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher);
+};
+
+} // namespace android
+
+#endif // PLAYLIST_FETCHER_H_
+
diff --git a/media/libstagefright/id3/Android.mk b/media/libstagefright/id3/Android.mk
index 80a1a3a..bf6f7bb 100644
--- a/media/libstagefright/id3/Android.mk
+++ b/media/libstagefright/id3/Android.mk
@@ -21,7 +21,7 @@
LOCAL_STATIC_LIBRARIES := \
libstagefright_id3
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE := testid3
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 22c2f5a..34d671a 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -30,12 +30,55 @@
static const size_t kMaxMetadataSize = 3 * 1024 * 1024;
+struct MemorySource : public DataSource {
+ MemorySource(const uint8_t *data, size_t size)
+ : mData(data),
+ mSize(size) {
+ }
+
+ virtual status_t initCheck() const {
+ return OK;
+ }
+
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size) {
+ off64_t available = (offset >= mSize) ? 0ll : mSize - offset;
+
+ size_t copy = (available > size) ? size : available;
+ memcpy(data, mData + offset, copy);
+
+ return copy;
+ }
+
+private:
+ const uint8_t *mData;
+ size_t mSize;
+
+ DISALLOW_EVIL_CONSTRUCTORS(MemorySource);
+};
+
ID3::ID3(const sp<DataSource> &source, bool ignoreV1)
: mIsValid(false),
mData(NULL),
mSize(0),
mFirstFrameOffset(0),
- mVersion(ID3_UNKNOWN) {
+ mVersion(ID3_UNKNOWN),
+ mRawSize(0) {
+ mIsValid = parseV2(source);
+
+ if (!mIsValid && !ignoreV1) {
+ mIsValid = parseV1(source);
+ }
+}
+
+ID3::ID3(const uint8_t *data, size_t size, bool ignoreV1)
+ : mIsValid(false),
+ mData(NULL),
+ mSize(0),
+ mFirstFrameOffset(0),
+ mVersion(ID3_UNKNOWN),
+ mRawSize(0) {
+ sp<MemorySource> source = new MemorySource(data, size);
+
mIsValid = parseV2(source);
if (!mIsValid && !ignoreV1) {
@@ -140,6 +183,7 @@
}
mSize = size;
+ mRawSize = mSize + sizeof(header);
if (source->readAt(sizeof(header), mData, mSize) != (ssize_t)mSize) {
free(mData);
@@ -313,17 +357,22 @@
}
if (flags & 2) {
- // Unsynchronization added.
+ // This file has "unsynchronization", so we have to replace occurrences
+ // of 0xff 0x00 with just 0xff in order to get the real data.
+ size_t readOffset = offset + 11;
+ size_t writeOffset = offset + 11;
for (size_t i = 0; i + 1 < dataSize; ++i) {
- if (mData[offset + 10 + i] == 0xff
- && mData[offset + 11 + i] == 0x00) {
- memmove(&mData[offset + 11 + i], &mData[offset + 12 + i],
- mSize - offset - 12 - i);
+ if (mData[readOffset - 1] == 0xff
+ && mData[readOffset] == 0x00) {
+ ++readOffset;
--mSize;
--dataSize;
}
+ mData[writeOffset++] = mData[readOffset++];
}
+ // move the remaining data following this frame
+ memmove(&mData[writeOffset], &mData[readOffset], oldSize - readOffset);
flags &= ~2;
}
@@ -505,7 +554,7 @@
int32_t i = n - 4;
while(--i >= 0 && *++frameData != 0) ;
int skipped = (frameData - mFrameData);
- if (skipped >= n) {
+ if (skipped >= (int)n) {
return;
}
n -= skipped;
diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h
index 3028f56..cca83ab 100644
--- a/media/libstagefright/include/ID3.h
+++ b/media/libstagefright/include/ID3.h
@@ -36,6 +36,7 @@
};
ID3(const sp<DataSource> &source, bool ignoreV1 = false);
+ ID3(const uint8_t *data, size_t size, bool ignoreV1 = false);
~ID3();
bool isValid() const;
@@ -71,6 +72,8 @@
Iterator &operator=(const Iterator &);
};
+ size_t rawSize() const { return mRawSize; }
+
private:
bool mIsValid;
uint8_t *mData;
@@ -78,6 +81,10 @@
size_t mFirstFrameOffset;
Version mVersion;
+ // size of the ID3 tag including header before any unsynchronization.
+ // only valid for IDV2+
+ size_t mRawSize;
+
bool parseV1(const sp<DataSource> &source);
bool parseV2(const sp<DataSource> &source);
void removeUnsynchronization();
diff --git a/media/libstagefright/include/LiveSession.h b/media/libstagefright/include/LiveSession.h
deleted file mode 100644
index db44a33..0000000
--- a/media/libstagefright/include/LiveSession.h
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef LIVE_SESSION_H_
-
-#define LIVE_SESSION_H_
-
-#include <media/stagefright/foundation/AHandler.h>
-
-#include <utils/String8.h>
-
-namespace android {
-
-struct ABuffer;
-struct DataSource;
-struct LiveDataSource;
-struct M3UParser;
-struct HTTPBase;
-
-struct LiveSession : public AHandler {
- enum Flags {
- // Don't log any URLs.
- kFlagIncognito = 1,
- };
- LiveSession(
- const sp<AMessage> ¬ify,
- uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
-
- sp<DataSource> getDataSource();
-
- void connect(
- const char *url,
- const KeyedVector<String8, String8> *headers = NULL);
-
- void disconnect();
-
- // Blocks until seek is complete.
- void seekTo(int64_t timeUs);
-
- status_t getDuration(int64_t *durationUs) const;
-
- bool isSeekable() const;
- bool hasDynamicDuration() const;
-
- // Posted notification's "what" field will carry one of the following:
- enum {
- kWhatPrepared,
- kWhatPreparationFailed,
- };
-
-protected:
- virtual ~LiveSession();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
- enum {
- kMaxNumQueuedFragments = 3,
- kMaxNumRetries = 5,
- };
-
- enum {
- kWhatConnect = 'conn',
- kWhatDisconnect = 'disc',
- kWhatMonitorQueue = 'moni',
- kWhatSeek = 'seek',
- };
-
- struct BandwidthItem {
- AString mURI;
- unsigned long mBandwidth;
- };
-
- sp<AMessage> mNotify;
- uint32_t mFlags;
- bool mUIDValid;
- uid_t mUID;
-
- bool mInPreparationPhase;
-
- sp<LiveDataSource> mDataSource;
-
- sp<HTTPBase> mHTTPDataSource;
-
- AString mMasterURL;
- KeyedVector<String8, String8> mExtraHeaders;
-
- Vector<BandwidthItem> mBandwidthItems;
-
- KeyedVector<AString, sp<ABuffer> > mAESKeyForURI;
-
- ssize_t mPrevBandwidthIndex;
- int64_t mLastPlaylistFetchTimeUs;
- sp<M3UParser> mPlaylist;
- int32_t mSeqNumber;
- int64_t mSeekTimeUs;
- int32_t mNumRetries;
- bool mStartOfPlayback;
-
- mutable Mutex mLock;
- Condition mCondition;
- int64_t mDurationUs;
- bool mDurationFixed; // Duration has been determined once and for all.
- bool mSeekDone;
- bool mDisconnectPending;
-
- int32_t mMonitorQueueGeneration;
-
- enum RefreshState {
- INITIAL_MINIMUM_RELOAD_DELAY,
- FIRST_UNCHANGED_RELOAD_ATTEMPT,
- SECOND_UNCHANGED_RELOAD_ATTEMPT,
- THIRD_UNCHANGED_RELOAD_ATTEMPT
- };
- RefreshState mRefreshState;
-
- uint8_t mPlaylistHash[16];
-
- void onConnect(const sp<AMessage> &msg);
- void onDisconnect();
- void onDownloadNext();
- void onMonitorQueue();
- void onSeek(const sp<AMessage> &msg);
-
- status_t fetchFile(
- const char *url, sp<ABuffer> *out,
- int64_t range_offset = 0, int64_t range_length = -1);
-
- sp<M3UParser> fetchPlaylist(const char *url, bool *unchanged);
- size_t getBandwidthIndex();
-
- status_t decryptBuffer(
- size_t playlistIndex, const sp<ABuffer> &buffer);
-
- void postMonitorQueue(int64_t delayUs = 0);
-
- bool timeToRefreshPlaylist(int64_t nowUs) const;
-
- static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *);
-
- // Returns the media time in us of the segment specified by seqNumber.
- // This is computed by summing the durations of all segments before it.
- int64_t getSegmentStartTimeUs(int32_t seqNumber) const;
-
- void signalEOS(status_t err);
-
- DISALLOW_EVIL_CONSTRUCTORS(LiveSession);
-};
-
-} // namespace android
-
-#endif // LIVE_SESSION_H_
diff --git a/media/libstagefright/include/MPEG2TSExtractor.h b/media/libstagefright/include/MPEG2TSExtractor.h
index fe74a42..c5e86a6 100644
--- a/media/libstagefright/include/MPEG2TSExtractor.h
+++ b/media/libstagefright/include/MPEG2TSExtractor.h
@@ -31,7 +31,6 @@
struct DataSource;
struct MPEG2TSSource;
struct String8;
-struct LiveSession;
struct MPEG2TSExtractor : public MediaExtractor {
MPEG2TSExtractor(const sp<DataSource> &source);
@@ -44,16 +43,12 @@
virtual uint32_t flags() const;
- void setLiveSession(const sp<LiveSession> &liveSession);
- void seekTo(int64_t seekTimeUs);
-
private:
friend struct MPEG2TSSource;
mutable Mutex mLock;
sp<DataSource> mDataSource;
- sp<LiveSession> mLiveSession;
sp<ATSParser> mParser;
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index 35eff96..bbec1c4 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -82,6 +82,7 @@
sp<DataSource> mDataSource;
status_t mInitCheck;
bool mHasVideo;
+ uint32_t mHeaderTimescale;
Track *mFirstTrack, *mLastTrack;
diff --git a/media/libstagefright/include/SoftVideoDecoderOMXComponent.h b/media/libstagefright/include/SoftVideoDecoderOMXComponent.h
new file mode 100644
index 0000000..d050fa6
--- /dev/null
+++ b/media/libstagefright/include/SoftVideoDecoderOMXComponent.h
@@ -0,0 +1,93 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SOFT_VIDEO_DECODER_OMX_COMPONENT_H_
+
+#define SOFT_VIDEO_DECODER_OMX_COMPONENT_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <media/IOMX.h>
+
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+#include <utils/Vector.h>
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof(*(a)))
+
+namespace android {
+
+struct SoftVideoDecoderOMXComponent : public SimpleSoftOMXComponent {
+ SoftVideoDecoderOMXComponent(
+ const char *name,
+ const char *componentRole,
+ OMX_VIDEO_CODINGTYPE codingType,
+ const CodecProfileLevel *profileLevels,
+ size_t numProfileLevels,
+ int32_t width,
+ int32_t height,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled);
+ virtual void onReset();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual OMX_ERRORTYPE getConfig(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ void initPorts(OMX_U32 numInputBuffers,
+ OMX_U32 inputBufferSize,
+ OMX_U32 numOutputBuffers,
+ const char *mimeType);
+
+ virtual void updatePortDefinitions();
+
+ enum {
+ kInputPortIndex = 0,
+ kOutputPortIndex = 1,
+ kMaxPortIndex = 1,
+ };
+
+ uint32_t mWidth, mHeight;
+ uint32_t mCropLeft, mCropTop, mCropWidth, mCropHeight;
+
+ enum {
+ NONE,
+ AWAITING_DISABLED,
+ AWAITING_ENABLED
+ } mOutputPortSettingsChange;
+
+private:
+ const char *mComponentRole;
+ OMX_VIDEO_CODINGTYPE mCodingType;
+ const CodecProfileLevel *mProfileLevels;
+ size_t mNumProfileLevels;
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftVideoDecoderOMXComponent);
+};
+
+} // namespace android
+
+#endif // SOFT_VIDEO_DECODER_OMX_COMPONENT_H_
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 3de3a61..3153c8b 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -32,9 +32,22 @@
AnotherPacketSource::AnotherPacketSource(const sp<MetaData> &meta)
: mIsAudio(false),
- mFormat(meta),
+ mFormat(NULL),
mLastQueuedTimeUs(0),
mEOSResult(OK) {
+ setFormat(meta);
+}
+
+void AnotherPacketSource::setFormat(const sp<MetaData> &meta) {
+ CHECK(mFormat == NULL);
+
+ mIsAudio = false;
+
+ if (meta == NULL) {
+ return;
+ }
+
+ mFormat = meta;
const char *mime;
CHECK(meta->findCString(kKeyMIMEType, &mime));
@@ -45,11 +58,6 @@
}
}
-void AnotherPacketSource::setFormat(const sp<MetaData> &meta) {
- CHECK(mFormat == NULL);
- mFormat = meta;
-}
-
AnotherPacketSource::~AnotherPacketSource() {
}
@@ -152,6 +160,15 @@
mCondition.signal();
}
+void AnotherPacketSource::clear() {
+ Mutex::Autolock autoLock(mLock);
+
+ mBuffers.clear();
+ mEOSResult = OK;
+
+ mFormat = NULL;
+}
+
void AnotherPacketSource::queueDiscontinuity(
ATSParser::DiscontinuityType type,
const sp<AMessage> &extra) {
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.h b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
index 1db4068..e16cf78 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.h
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
@@ -41,6 +41,8 @@
virtual status_t read(
MediaBuffer **buffer, const ReadOptions *options = NULL);
+ void clear();
+
bool hasBufferAvailable(status_t *finalResult);
// Returns the difference between the last and the first queued
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index e1589b4..d449c34 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -19,7 +19,6 @@
#include <utils/Log.h>
#include "include/MPEG2TSExtractor.h"
-#include "include/LiveSession.h"
#include "include/NuCachedSource2.h"
#include <media/stagefright/foundation/ADebug.h>
@@ -79,15 +78,7 @@
}
sp<MetaData> MPEG2TSSource::getFormat() {
- sp<MetaData> meta = mImpl->getFormat();
-
- int64_t durationUs;
- if (mExtractor->mLiveSession != NULL
- && mExtractor->mLiveSession->getDuration(&durationUs) == OK) {
- meta->setInt64(kKeyDuration, durationUs);
- }
-
- return meta;
+ return mImpl->getFormat();
}
status_t MPEG2TSSource::read(
@@ -97,7 +88,7 @@
int64_t seekTimeUs;
ReadOptions::SeekMode seekMode;
if (mSeekable && options && options->getSeekTo(&seekTimeUs, &seekMode)) {
- mExtractor->seekTo(seekTimeUs);
+ return ERROR_UNSUPPORTED;
}
status_t finalResult;
@@ -216,32 +207,8 @@
return mParser->feedTSPacket(packet, kTSPacketSize);
}
-void MPEG2TSExtractor::setLiveSession(const sp<LiveSession> &liveSession) {
- Mutex::Autolock autoLock(mLock);
-
- mLiveSession = liveSession;
-}
-
-void MPEG2TSExtractor::seekTo(int64_t seekTimeUs) {
- Mutex::Autolock autoLock(mLock);
-
- if (mLiveSession == NULL) {
- return;
- }
-
- mLiveSession->seekTo(seekTimeUs);
-}
-
uint32_t MPEG2TSExtractor::flags() const {
- Mutex::Autolock autoLock(mLock);
-
- uint32_t flags = CAN_PAUSE;
-
- if (mLiveSession != NULL && mLiveSession->isSeekable()) {
- flags |= CAN_SEEK_FORWARD | CAN_SEEK_BACKWARD | CAN_SEEK;
- }
-
- return flags;
+ return CAN_PAUSE;
}
////////////////////////////////////////////////////////////////////////////////
diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk
index a8b4939..cd912e7 100644
--- a/media/libstagefright/omx/Android.mk
+++ b/media/libstagefright/omx/Android.mk
@@ -9,6 +9,7 @@
SimpleSoftOMXComponent.cpp \
SoftOMXComponent.cpp \
SoftOMXPlugin.cpp \
+ SoftVideoDecoderOMXComponent.cpp \
LOCAL_C_INCLUDES += \
$(TOP)/frameworks/av/media/libstagefright \
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index ef27879..b3a8463 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -206,24 +206,15 @@
// Find matching entry in our cached copy of the BufferQueue slots.
// If we find a match, release that slot. If we don't, the BufferQueue
// has dropped that GraphicBuffer, and there's nothing for us to release.
- //
- // (We could store "id" in CodecBuffer and avoid the slot search.)
- int id;
- for (id = 0; id < BufferQueue::NUM_BUFFER_SLOTS; id++) {
- if (mBufferSlot[id] == NULL) {
- continue;
- }
+ int id = codecBuffer.mBuf;
+ if (mBufferSlot[id] != NULL &&
+ mBufferSlot[id]->handle == codecBuffer.mGraphicBuffer->handle) {
+ ALOGV("cbi %d matches bq slot %d, handle=%p",
+ cbi, id, mBufferSlot[id]->handle);
- if (mBufferSlot[id]->handle == codecBuffer.mGraphicBuffer->handle) {
- ALOGV("cbi %d matches bq slot %d, handle=%p",
- cbi, id, mBufferSlot[id]->handle);
-
- mBufferQueue->releaseBuffer(id, EGL_NO_DISPLAY, EGL_NO_SYNC_KHR,
- Fence::NO_FENCE);
- break;
- }
- }
- if (id == BufferQueue::NUM_BUFFER_SLOTS) {
+ mBufferQueue->releaseBuffer(id, codecBuffer.mFrameNumber,
+ EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
+ } else {
ALOGV("codecBufferEmptied: no match for emptied buffer in cbi %d",
cbi);
}
@@ -287,11 +278,11 @@
mBufferSlot[item.mBuf] = item.mGraphicBuffer;
}
- err = submitBuffer_l(mBufferSlot[item.mBuf], item.mTimestamp / 1000, cbi);
+ err = submitBuffer_l(item, cbi);
if (err != OK) {
ALOGV("submitBuffer_l failed, releasing bq buf %d", item.mBuf);
- mBufferQueue->releaseBuffer(item.mBuf, EGL_NO_DISPLAY,
- EGL_NO_SYNC_KHR, Fence::NO_FENCE);
+ mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
+ EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
} else {
ALOGV("buffer submitted (bq %d, cbi %d)", item.mBuf, cbi);
}
@@ -326,11 +317,13 @@
return OK;
}
-status_t GraphicBufferSource::submitBuffer_l(sp<GraphicBuffer>& graphicBuffer,
- int64_t timestampUsec, int cbi) {
+status_t GraphicBufferSource::submitBuffer_l(
+ const BufferQueue::BufferItem &item, int cbi) {
ALOGV("submitBuffer_l cbi=%d", cbi);
CodecBuffer& codecBuffer(mCodecBuffers.editItemAt(cbi));
- codecBuffer.mGraphicBuffer = graphicBuffer;
+ codecBuffer.mGraphicBuffer = mBufferSlot[item.mBuf];
+ codecBuffer.mBuf = item.mBuf;
+ codecBuffer.mFrameNumber = item.mFrameNumber;
OMX_BUFFERHEADERTYPE* header = codecBuffer.mHeader;
CHECK(header->nAllocLen >= 4 + sizeof(buffer_handle_t));
@@ -342,7 +335,7 @@
status_t err = mNodeInstance->emptyDirectBuffer(header, 0,
4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME,
- timestampUsec);
+ item.mTimestamp / 1000);
if (err != OK) {
ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err);
codecBuffer.mGraphicBuffer = NULL;
@@ -431,8 +424,8 @@
BufferQueue::BufferItem item;
status_t err = mBufferQueue->acquireBuffer(&item);
if (err == OK) {
- mBufferQueue->releaseBuffer(item.mBuf, EGL_NO_DISPLAY,
- EGL_NO_SYNC_KHR, item.mFence);
+ mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
+ EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, item.mFence);
}
return;
}
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index 562d342..8c6b470 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -104,6 +104,13 @@
// (mGraphicBuffer == NULL) or in use by the codec.
struct CodecBuffer {
OMX_BUFFERHEADERTYPE* mHeader;
+
+ // buffer producer's frame-number for buffer
+ uint64_t mFrameNumber;
+
+ // buffer producer's buffer slot for buffer
+ int mBuf;
+
sp<GraphicBuffer> mGraphicBuffer;
};
@@ -130,8 +137,7 @@
// Marks the mCodecBuffers entry as in-use, copies the GraphicBuffer
// reference into the codec buffer, and submits the data to the codec.
- status_t submitBuffer_l(sp<GraphicBuffer>& graphicBuffer,
- int64_t timestampUsec, int cbi);
+ status_t submitBuffer_l(const BufferQueue::BufferItem &item, int cbi);
// Submits an empty buffer, with the EOS flag set. Returns without
// doing anything if we don't have a codec buffer available.
diff --git a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
new file mode 100644
index 0000000..08a3d42
--- /dev/null
+++ b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
@@ -0,0 +1,290 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftVideoDecoderOMXComponent"
+#include <utils/Log.h>
+
+#include "include/SoftVideoDecoderOMXComponent.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaDefs.h>
+
+namespace android {
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftVideoDecoderOMXComponent::SoftVideoDecoderOMXComponent(
+ const char *name,
+ const char *componentRole,
+ OMX_VIDEO_CODINGTYPE codingType,
+ const CodecProfileLevel *profileLevels,
+ size_t numProfileLevels,
+ int32_t width,
+ int32_t height,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mWidth(width),
+ mHeight(height),
+ mCropLeft(0),
+ mCropTop(0),
+ mCropWidth(width),
+ mCropHeight(height),
+ mOutputPortSettingsChange(NONE),
+ mComponentRole(componentRole),
+ mCodingType(codingType),
+ mProfileLevels(profileLevels),
+ mNumProfileLevels(numProfileLevels) {
+}
+
+void SoftVideoDecoderOMXComponent::initPorts(
+ OMX_U32 numInputBuffers,
+ OMX_U32 inputBufferSize,
+ OMX_U32 numOutputBuffers,
+ const char *mimeType) {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = kInputPortIndex;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = numInputBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = inputBufferSize;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainVideo;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.video.cMIMEType = const_cast<char *>(mimeType);
+ def.format.video.pNativeRender = NULL;
+ /* size is initialized in updatePortDefinitions() */
+ def.format.video.nBitrate = 0;
+ def.format.video.xFramerate = 0;
+ def.format.video.bFlagErrorConcealment = OMX_FALSE;
+ def.format.video.eCompressionFormat = mCodingType;
+ def.format.video.eColorFormat = OMX_COLOR_FormatUnused;
+ def.format.video.pNativeWindow = NULL;
+
+ addPort(def);
+
+ def.nPortIndex = kOutputPortIndex;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = numOutputBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainVideo;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.video.cMIMEType = const_cast<char *>("video/raw");
+ def.format.video.pNativeRender = NULL;
+ /* size is initialized in updatePortDefinitions() */
+ def.format.video.nBitrate = 0;
+ def.format.video.xFramerate = 0;
+ def.format.video.bFlagErrorConcealment = OMX_FALSE;
+ def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused;
+ def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar;
+ def.format.video.pNativeWindow = NULL;
+
+ addPort(def);
+
+ updatePortDefinitions();
+}
+
+void SoftVideoDecoderOMXComponent::updatePortDefinitions() {
+ OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(kInputPortIndex)->mDef;
+ def->format.video.nFrameWidth = mWidth;
+ def->format.video.nFrameHeight = mHeight;
+ def->format.video.nStride = def->format.video.nFrameWidth;
+ def->format.video.nSliceHeight = def->format.video.nFrameHeight;
+
+ def = &editPortInfo(kOutputPortIndex)->mDef;
+ def->format.video.nFrameWidth = mWidth;
+ def->format.video.nFrameHeight = mHeight;
+ def->format.video.nStride = def->format.video.nFrameWidth;
+ def->format.video.nSliceHeight = def->format.video.nFrameHeight;
+
+ def->nBufferSize =
+ (def->format.video.nFrameWidth *
+ def->format.video.nFrameHeight * 3) / 2;
+
+ mCropLeft = 0;
+ mCropTop = 0;
+ mCropWidth = mWidth;
+ mCropHeight = mHeight;
+}
+
+OMX_ERRORTYPE SoftVideoDecoderOMXComponent::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamVideoPortFormat:
+ {
+ OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > kMaxPortIndex) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex != 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if (formatParams->nPortIndex == kInputPortIndex) {
+ formatParams->eCompressionFormat = mCodingType;
+ formatParams->eColorFormat = OMX_COLOR_FormatUnused;
+ formatParams->xFramerate = 0;
+ } else {
+ CHECK_EQ(formatParams->nPortIndex, 1u);
+
+ formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused;
+ formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar;
+ formatParams->xFramerate = 0;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamVideoProfileLevelQuerySupported:
+ {
+ OMX_VIDEO_PARAM_PROFILELEVELTYPE *profileLevel =
+ (OMX_VIDEO_PARAM_PROFILELEVELTYPE *) params;
+
+ if (profileLevel->nPortIndex != kInputPortIndex) {
+ ALOGE("Invalid port index: %ld", profileLevel->nPortIndex);
+ return OMX_ErrorUnsupportedIndex;
+ }
+
+ if (index >= mNumProfileLevels) {
+ return OMX_ErrorNoMore;
+ }
+
+ profileLevel->eProfile = mProfileLevels[index].mProfile;
+ profileLevel->eLevel = mProfileLevels[index].mLevel;
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftVideoDecoderOMXComponent::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ mComponentRole,
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamVideoPortFormat:
+ {
+ OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > kMaxPortIndex) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex != 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftVideoDecoderOMXComponent::getConfig(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexConfigCommonOutputCrop:
+ {
+ OMX_CONFIG_RECTTYPE *rectParams = (OMX_CONFIG_RECTTYPE *)params;
+
+ if (rectParams->nPortIndex != kOutputPortIndex) {
+ return OMX_ErrorUndefined;
+ }
+
+ rectParams->nLeft = mCropLeft;
+ rectParams->nTop = mCropTop;
+ rectParams->nWidth = mCropWidth;
+ rectParams->nHeight = mCropHeight;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return OMX_ErrorUnsupportedIndex;
+ }
+}
+
+void SoftVideoDecoderOMXComponent::onReset() {
+ mOutputPortSettingsChange = NONE;
+}
+
+void SoftVideoDecoderOMXComponent::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
+ if (portIndex != kOutputPortIndex) {
+ return;
+ }
+
+ switch (mOutputPortSettingsChange) {
+ case NONE:
+ break;
+
+ case AWAITING_DISABLED:
+ {
+ CHECK(!enabled);
+ mOutputPortSettingsChange = AWAITING_ENABLED;
+ break;
+ }
+
+ default:
+ {
+ CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
+ CHECK(enabled);
+ mOutputPortSettingsChange = NONE;
+ break;
+ }
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk
index 9e2724d..e77c69c 100644
--- a/media/libstagefright/rtsp/Android.mk
+++ b/media/libstagefright/rtsp/Android.mk
@@ -51,7 +51,7 @@
LOCAL_CFLAGS += -Wno-multichar
-LOCAL_MODULE_TAGS := debug
+LOCAL_MODULE_TAGS := optional
LOCAL_MODULE:= rtp_test
diff --git a/media/libstagefright/wifi-display/Android.mk b/media/libstagefright/wifi-display/Android.mk
index 061ae89..404b41e 100644
--- a/media/libstagefright/wifi-display/Android.mk
+++ b/media/libstagefright/wifi-display/Android.mk
@@ -4,10 +4,17 @@
LOCAL_SRC_FILES:= \
ANetworkSession.cpp \
+ MediaReceiver.cpp \
MediaSender.cpp \
Parameters.cpp \
ParsedMessage.cpp \
+ rtp/RTPAssembler.cpp \
+ rtp/RTPReceiver.cpp \
rtp/RTPSender.cpp \
+ sink/DirectRenderer.cpp \
+ sink/WifiDisplaySink.cpp \
+ SNTPClient.cpp \
+ TimeSyncer.cpp \
source/Converter.cpp \
source/MediaPuller.cpp \
source/PlaybackSession.cpp \
@@ -57,6 +64,67 @@
LOCAL_MODULE:= wfd
-LOCAL_MODULE_TAGS := debug
+include $(BUILD_EXECUTABLE)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ udptest.cpp \
+
+LOCAL_SHARED_LIBRARIES:= \
+ libbinder \
+ libgui \
+ libmedia \
+ libstagefright \
+ libstagefright_foundation \
+ libstagefright_wfd \
+ libutils \
+ liblog \
+
+LOCAL_MODULE:= udptest
+
+include $(BUILD_EXECUTABLE)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ rtptest.cpp \
+
+LOCAL_SHARED_LIBRARIES:= \
+ libbinder \
+ libgui \
+ libmedia \
+ libstagefright \
+ libstagefright_foundation \
+ libstagefright_wfd \
+ libutils \
+ liblog \
+
+LOCAL_MODULE:= rtptest
+
+include $(BUILD_EXECUTABLE)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ nettest.cpp \
+
+LOCAL_SHARED_LIBRARIES:= \
+ libbinder \
+ libgui \
+ libmedia \
+ libstagefright \
+ libstagefright_foundation \
+ libstagefright_wfd \
+ libutils \
+ liblog \
+
+LOCAL_MODULE:= nettest
include $(BUILD_EXECUTABLE)
diff --git a/media/libstagefright/wifi-display/MediaReceiver.cpp b/media/libstagefright/wifi-display/MediaReceiver.cpp
new file mode 100644
index 0000000..364acb9
--- /dev/null
+++ b/media/libstagefright/wifi-display/MediaReceiver.cpp
@@ -0,0 +1,328 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaReceiver"
+#include <utils/Log.h>
+
+#include "MediaReceiver.h"
+
+#include "ANetworkSession.h"
+#include "AnotherPacketSource.h"
+#include "rtp/RTPReceiver.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+MediaReceiver::MediaReceiver(
+ const sp<ANetworkSession> &netSession,
+ const sp<AMessage> ¬ify)
+ : mNetSession(netSession),
+ mNotify(notify),
+ mMode(MODE_UNDEFINED),
+ mGeneration(0),
+ mInitStatus(OK),
+ mInitDoneCount(0) {
+}
+
+MediaReceiver::~MediaReceiver() {
+}
+
+ssize_t MediaReceiver::addTrack(
+ RTPReceiver::TransportMode rtpMode,
+ RTPReceiver::TransportMode rtcpMode,
+ int32_t *localRTPPort) {
+ if (mMode != MODE_UNDEFINED) {
+ return INVALID_OPERATION;
+ }
+
+ size_t trackIndex = mTrackInfos.size();
+
+ TrackInfo info;
+
+ sp<AMessage> notify = new AMessage(kWhatReceiverNotify, id());
+ notify->setInt32("generation", mGeneration);
+ notify->setSize("trackIndex", trackIndex);
+
+ info.mReceiver = new RTPReceiver(mNetSession, notify);
+ looper()->registerHandler(info.mReceiver);
+
+ info.mReceiver->registerPacketType(
+ 33, RTPReceiver::PACKETIZATION_TRANSPORT_STREAM);
+
+ info.mReceiver->registerPacketType(
+ 96, RTPReceiver::PACKETIZATION_AAC);
+
+ info.mReceiver->registerPacketType(
+ 97, RTPReceiver::PACKETIZATION_H264);
+
+ status_t err = info.mReceiver->initAsync(
+ rtpMode,
+ rtcpMode,
+ localRTPPort);
+
+ if (err != OK) {
+ looper()->unregisterHandler(info.mReceiver->id());
+ info.mReceiver.clear();
+
+ return err;
+ }
+
+ mTrackInfos.push_back(info);
+
+ return trackIndex;
+}
+
+status_t MediaReceiver::connectTrack(
+ size_t trackIndex,
+ const char *remoteHost,
+ int32_t remoteRTPPort,
+ int32_t remoteRTCPPort) {
+ if (trackIndex >= mTrackInfos.size()) {
+ return -ERANGE;
+ }
+
+ TrackInfo *info = &mTrackInfos.editItemAt(trackIndex);
+ return info->mReceiver->connect(remoteHost, remoteRTPPort, remoteRTCPPort);
+}
+
+status_t MediaReceiver::initAsync(Mode mode) {
+ if ((mode == MODE_TRANSPORT_STREAM || mode == MODE_TRANSPORT_STREAM_RAW)
+ && mTrackInfos.size() > 1) {
+ return INVALID_OPERATION;
+ }
+
+ sp<AMessage> msg = new AMessage(kWhatInit, id());
+ msg->setInt32("mode", mode);
+ msg->post();
+
+ return OK;
+}
+
+void MediaReceiver::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatInit:
+ {
+ int32_t mode;
+ CHECK(msg->findInt32("mode", &mode));
+
+ CHECK_EQ(mMode, MODE_UNDEFINED);
+ mMode = (Mode)mode;
+
+ if (mInitStatus != OK || mInitDoneCount == mTrackInfos.size()) {
+ notifyInitDone(mInitStatus);
+ }
+
+ mTSParser = new ATSParser(
+ ATSParser::ALIGNED_VIDEO_DATA
+ | ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
+
+ mFormatKnownMask = 0;
+ break;
+ }
+
+ case kWhatReceiverNotify:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+ if (generation != mGeneration) {
+ break;
+ }
+
+ onReceiverNotify(msg);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void MediaReceiver::onReceiverNotify(const sp<AMessage> &msg) {
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case RTPReceiver::kWhatInitDone:
+ {
+ ++mInitDoneCount;
+
+ int32_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ if (err != OK) {
+ mInitStatus = err;
+ ++mGeneration;
+ }
+
+ if (mMode != MODE_UNDEFINED) {
+ if (mInitStatus != OK || mInitDoneCount == mTrackInfos.size()) {
+ notifyInitDone(mInitStatus);
+ }
+ }
+ break;
+ }
+
+ case RTPReceiver::kWhatError:
+ {
+ int32_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ notifyError(err);
+ break;
+ }
+
+ case RTPReceiver::kWhatAccessUnit:
+ {
+ size_t trackIndex;
+ CHECK(msg->findSize("trackIndex", &trackIndex));
+
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("accessUnit", &accessUnit));
+
+ int32_t followsDiscontinuity;
+ if (!msg->findInt32(
+ "followsDiscontinuity", &followsDiscontinuity)) {
+ followsDiscontinuity = 0;
+ }
+
+ if (mMode == MODE_TRANSPORT_STREAM) {
+ if (followsDiscontinuity) {
+ mTSParser->signalDiscontinuity(
+ ATSParser::DISCONTINUITY_TIME, NULL /* extra */);
+ }
+
+ for (size_t offset = 0;
+ offset < accessUnit->size(); offset += 188) {
+ status_t err = mTSParser->feedTSPacket(
+ accessUnit->data() + offset, 188);
+
+ if (err != OK) {
+ notifyError(err);
+ break;
+ }
+ }
+
+ drainPackets(0 /* trackIndex */, ATSParser::VIDEO);
+ drainPackets(1 /* trackIndex */, ATSParser::AUDIO);
+ } else {
+ postAccessUnit(trackIndex, accessUnit, NULL);
+ }
+ break;
+ }
+
+ case RTPReceiver::kWhatPacketLost:
+ {
+ notifyPacketLost();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void MediaReceiver::drainPackets(
+ size_t trackIndex, ATSParser::SourceType type) {
+ sp<AnotherPacketSource> source =
+ static_cast<AnotherPacketSource *>(
+ mTSParser->getSource(type).get());
+
+ if (source == NULL) {
+ return;
+ }
+
+ sp<AMessage> format;
+ if (!(mFormatKnownMask & (1ul << trackIndex))) {
+ sp<MetaData> meta = source->getFormat();
+ CHECK(meta != NULL);
+
+ CHECK_EQ((status_t)OK, convertMetaDataToMessage(meta, &format));
+
+ mFormatKnownMask |= 1ul << trackIndex;
+ }
+
+ status_t finalResult;
+ while (source->hasBufferAvailable(&finalResult)) {
+ sp<ABuffer> accessUnit;
+ status_t err = source->dequeueAccessUnit(&accessUnit);
+ if (err == OK) {
+ postAccessUnit(trackIndex, accessUnit, format);
+ format.clear();
+ } else if (err != INFO_DISCONTINUITY) {
+ notifyError(err);
+ }
+ }
+
+ if (finalResult != OK) {
+ notifyError(finalResult);
+ }
+}
+
+void MediaReceiver::notifyInitDone(status_t err) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatInitDone);
+ notify->setInt32("err", err);
+ notify->post();
+}
+
+void MediaReceiver::notifyError(status_t err) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatError);
+ notify->setInt32("err", err);
+ notify->post();
+}
+
+void MediaReceiver::notifyPacketLost() {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatPacketLost);
+ notify->post();
+}
+
+void MediaReceiver::postAccessUnit(
+ size_t trackIndex,
+ const sp<ABuffer> &accessUnit,
+ const sp<AMessage> &format) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatAccessUnit);
+ notify->setSize("trackIndex", trackIndex);
+ notify->setBuffer("accessUnit", accessUnit);
+
+ if (format != NULL) {
+ notify->setMessage("format", format);
+ }
+
+ notify->post();
+}
+
+status_t MediaReceiver::informSender(
+ size_t trackIndex, const sp<AMessage> ¶ms) {
+ if (trackIndex >= mTrackInfos.size()) {
+ return -ERANGE;
+ }
+
+ TrackInfo *info = &mTrackInfos.editItemAt(trackIndex);
+ return info->mReceiver->informSender(params);
+}
+
+} // namespace android
+
+
diff --git a/media/libstagefright/wifi-display/MediaReceiver.h b/media/libstagefright/wifi-display/MediaReceiver.h
new file mode 100644
index 0000000..afbb407
--- /dev/null
+++ b/media/libstagefright/wifi-display/MediaReceiver.h
@@ -0,0 +1,111 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/stagefright/foundation/AHandler.h>
+
+#include "ATSParser.h"
+#include "rtp/RTPReceiver.h"
+
+namespace android {
+
+struct ABuffer;
+struct ANetworkSession;
+struct AMessage;
+struct ATSParser;
+
+// This class facilitates receiving of media data for one or more tracks
+// over RTP. Either a 1:1 track to RTP channel mapping is used or a single
+// RTP channel provides the data for a transport stream that is consequently
+// demuxed and its track's data provided to the observer.
+struct MediaReceiver : public AHandler {
+ enum {
+ kWhatInitDone,
+ kWhatError,
+ kWhatAccessUnit,
+ kWhatPacketLost,
+ };
+
+ MediaReceiver(
+ const sp<ANetworkSession> &netSession,
+ const sp<AMessage> ¬ify);
+
+ ssize_t addTrack(
+ RTPReceiver::TransportMode rtpMode,
+ RTPReceiver::TransportMode rtcpMode,
+ int32_t *localRTPPort);
+
+ status_t connectTrack(
+ size_t trackIndex,
+ const char *remoteHost,
+ int32_t remoteRTPPort,
+ int32_t remoteRTCPPort);
+
+ enum Mode {
+ MODE_UNDEFINED,
+ MODE_TRANSPORT_STREAM,
+ MODE_TRANSPORT_STREAM_RAW,
+ MODE_ELEMENTARY_STREAMS,
+ };
+ status_t initAsync(Mode mode);
+
+ status_t informSender(size_t trackIndex, const sp<AMessage> ¶ms);
+
+protected:
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+ virtual ~MediaReceiver();
+
+private:
+ enum {
+ kWhatInit,
+ kWhatReceiverNotify,
+ };
+
+ struct TrackInfo {
+ sp<RTPReceiver> mReceiver;
+ };
+
+ sp<ANetworkSession> mNetSession;
+ sp<AMessage> mNotify;
+
+ Mode mMode;
+ int32_t mGeneration;
+
+ Vector<TrackInfo> mTrackInfos;
+
+ status_t mInitStatus;
+ size_t mInitDoneCount;
+
+ sp<ATSParser> mTSParser;
+ uint32_t mFormatKnownMask;
+
+ void onReceiverNotify(const sp<AMessage> &msg);
+
+ void drainPackets(size_t trackIndex, ATSParser::SourceType type);
+
+ void notifyInitDone(status_t err);
+ void notifyError(status_t err);
+ void notifyPacketLost();
+
+ void postAccessUnit(
+ size_t trackIndex,
+ const sp<ABuffer> &accessUnit,
+ const sp<AMessage> &format);
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaReceiver);
+};
+
+} // namespace android
+
diff --git a/media/libstagefright/wifi-display/MediaSender.cpp b/media/libstagefright/wifi-display/MediaSender.cpp
index 8a3566f..a230cd8 100644
--- a/media/libstagefright/wifi-display/MediaSender.cpp
+++ b/media/libstagefright/wifi-display/MediaSender.cpp
@@ -27,9 +27,11 @@
#include "include/avc_utils.h"
#include <media/IHDCP.h>
+#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
+#include <ui/GraphicBuffer.h>
namespace android {
@@ -341,6 +343,22 @@
break;
}
+ case kWhatInformSender:
+ {
+ int64_t avgLatencyUs;
+ CHECK(msg->findInt64("avgLatencyUs", &avgLatencyUs));
+
+ int64_t maxLatencyUs;
+ CHECK(msg->findInt64("maxLatencyUs", &maxLatencyUs));
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatInformSender);
+ notify->setInt64("avgLatencyUs", avgLatencyUs);
+ notify->setInt64("maxLatencyUs", maxLatencyUs);
+ notify->post();
+ break;
+ }
+
default:
TRESPASS();
}
@@ -392,11 +410,36 @@
info.mPacketizerTrackIndex, accessUnit);
}
- status_t err = mHDCP->encrypt(
- accessUnit->data(), accessUnit->size(),
- trackIndex /* streamCTR */,
- &inputCTR,
- accessUnit->data());
+ status_t err;
+ native_handle_t* handle;
+ if (accessUnit->meta()->findPointer("handle", (void**)&handle)
+ && handle != NULL) {
+ int32_t rangeLength, rangeOffset;
+ sp<AMessage> notify;
+ CHECK(accessUnit->meta()->findInt32("rangeOffset", &rangeOffset));
+ CHECK(accessUnit->meta()->findInt32("rangeLength", &rangeLength));
+ CHECK(accessUnit->meta()->findMessage("notify", ¬ify)
+ && notify != NULL);
+ CHECK_GE(accessUnit->size(), rangeLength);
+
+ sp<GraphicBuffer> grbuf(new GraphicBuffer(
+ rangeOffset + rangeLength, 1, HAL_PIXEL_FORMAT_Y8,
+ GRALLOC_USAGE_HW_VIDEO_ENCODER, rangeOffset + rangeLength,
+ handle, false));
+
+ err = mHDCP->encryptNative(
+ grbuf, rangeOffset, rangeLength,
+ trackIndex /* streamCTR */,
+ &inputCTR,
+ accessUnit->data());
+ notify->post();
+ } else {
+ err = mHDCP->encrypt(
+ accessUnit->data(), accessUnit->size(),
+ trackIndex /* streamCTR */,
+ &inputCTR,
+ accessUnit->data());
+ }
if (err != OK) {
ALOGE("Failed to HDCP-encrypt media data (err %d)",
diff --git a/media/libstagefright/wifi-display/MediaSender.h b/media/libstagefright/wifi-display/MediaSender.h
index 64722c5..04538ea 100644
--- a/media/libstagefright/wifi-display/MediaSender.h
+++ b/media/libstagefright/wifi-display/MediaSender.h
@@ -43,6 +43,7 @@
kWhatInitDone,
kWhatError,
kWhatNetworkStall,
+ kWhatInformSender,
};
MediaSender(
diff --git a/media/libstagefright/wifi-display/SNTPClient.cpp b/media/libstagefright/wifi-display/SNTPClient.cpp
new file mode 100644
index 0000000..5c0af6a
--- /dev/null
+++ b/media/libstagefright/wifi-display/SNTPClient.cpp
@@ -0,0 +1,174 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "SNTPClient.h"
+
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/Utils.h>
+
+#include <arpa/inet.h>
+#include <netdb.h>
+#include <netinet/in.h>
+#include <sys/socket.h>
+#include <unistd.h>
+
+namespace android {
+
+SNTPClient::SNTPClient() {
+}
+
+status_t SNTPClient::requestTime(const char *host) {
+ struct hostent *ent;
+ int64_t requestTimeNTP, requestTimeUs;
+ ssize_t n;
+ int64_t responseTimeUs, responseTimeNTP;
+ int64_t originateTimeNTP, receiveTimeNTP, transmitTimeNTP;
+ int64_t roundTripTimeNTP, clockOffsetNTP;
+
+ status_t err = UNKNOWN_ERROR;
+
+ int s = socket(AF_INET, SOCK_DGRAM, 0);
+
+ if (s < 0) {
+ err = -errno;
+
+ goto bail;
+ }
+
+ ent = gethostbyname(host);
+
+ if (ent == NULL) {
+ err = -ENOENT;
+ goto bail2;
+ }
+
+ struct sockaddr_in hostAddr;
+ memset(hostAddr.sin_zero, 0, sizeof(hostAddr.sin_zero));
+ hostAddr.sin_family = AF_INET;
+ hostAddr.sin_port = htons(kNTPPort);
+ hostAddr.sin_addr.s_addr = *(in_addr_t *)ent->h_addr;
+
+ uint8_t packet[kNTPPacketSize];
+ memset(packet, 0, sizeof(packet));
+
+ packet[0] = kNTPModeClient | (kNTPVersion << 3);
+
+ requestTimeNTP = getNowNTP();
+ requestTimeUs = ALooper::GetNowUs();
+ writeTimeStamp(&packet[kNTPTransmitTimeOffset], requestTimeNTP);
+
+ n = sendto(
+ s, packet, sizeof(packet), 0,
+ (const struct sockaddr *)&hostAddr, sizeof(hostAddr));
+
+ if (n < 0) {
+ err = -errno;
+ goto bail2;
+ }
+
+ memset(packet, 0, sizeof(packet));
+
+ do {
+ n = recv(s, packet, sizeof(packet), 0);
+ } while (n < 0 && errno == EINTR);
+
+ if (n < 0) {
+ err = -errno;
+ goto bail2;
+ }
+
+ responseTimeUs = ALooper::GetNowUs();
+
+ responseTimeNTP = requestTimeNTP + makeNTP(responseTimeUs - requestTimeUs);
+
+ originateTimeNTP = readTimeStamp(&packet[kNTPOriginateTimeOffset]);
+ receiveTimeNTP = readTimeStamp(&packet[kNTPReceiveTimeOffset]);
+ transmitTimeNTP = readTimeStamp(&packet[kNTPTransmitTimeOffset]);
+
+ roundTripTimeNTP =
+ makeNTP(responseTimeUs - requestTimeUs)
+ - (transmitTimeNTP - receiveTimeNTP);
+
+ clockOffsetNTP =
+ ((receiveTimeNTP - originateTimeNTP)
+ + (transmitTimeNTP - responseTimeNTP)) / 2;
+
+ mTimeReferenceNTP = responseTimeNTP + clockOffsetNTP;
+ mTimeReferenceUs = responseTimeUs;
+ mRoundTripTimeNTP = roundTripTimeNTP;
+
+ err = OK;
+
+bail2:
+ close(s);
+ s = -1;
+
+bail:
+ return err;
+}
+
+int64_t SNTPClient::adjustTimeUs(int64_t timeUs) const {
+ uint64_t nowNTP =
+ mTimeReferenceNTP + makeNTP(timeUs - mTimeReferenceUs);
+
+ int64_t nowUs =
+ (nowNTP >> 32) * 1000000ll
+ + ((nowNTP & 0xffffffff) * 1000000ll) / (1ll << 32);
+
+ nowUs -= ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll;
+
+ return nowUs;
+}
+
+// static
+void SNTPClient::writeTimeStamp(uint8_t *dst, uint64_t ntpTime) {
+ *dst++ = (ntpTime >> 56) & 0xff;
+ *dst++ = (ntpTime >> 48) & 0xff;
+ *dst++ = (ntpTime >> 40) & 0xff;
+ *dst++ = (ntpTime >> 32) & 0xff;
+ *dst++ = (ntpTime >> 24) & 0xff;
+ *dst++ = (ntpTime >> 16) & 0xff;
+ *dst++ = (ntpTime >> 8) & 0xff;
+ *dst++ = ntpTime & 0xff;
+}
+
+// static
+uint64_t SNTPClient::readTimeStamp(const uint8_t *dst) {
+ return U64_AT(dst);
+}
+
+// static
+uint64_t SNTPClient::getNowNTP() {
+ struct timeval tv;
+ gettimeofday(&tv, NULL /* time zone */);
+
+ uint64_t nowUs = tv.tv_sec * 1000000ll + tv.tv_usec;
+
+ nowUs += ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll;
+
+ return makeNTP(nowUs);
+}
+
+// static
+uint64_t SNTPClient::makeNTP(uint64_t deltaUs) {
+ uint64_t hi = deltaUs / 1000000ll;
+ uint64_t lo = ((1ll << 32) * (deltaUs % 1000000ll)) / 1000000ll;
+
+ return (hi << 32) | lo;
+}
+
+} // namespace android
+
diff --git a/media/libstagefright/wifi-display/SNTPClient.h b/media/libstagefright/wifi-display/SNTPClient.h
new file mode 100644
index 0000000..967d1fc
--- /dev/null
+++ b/media/libstagefright/wifi-display/SNTPClient.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SNTP_CLIENT_H_
+
+#define SNTP_CLIENT_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+// Implementation of the SNTP (Simple Network Time Protocol)
+struct SNTPClient {
+ SNTPClient();
+
+ status_t requestTime(const char *host);
+
+ // given a time obtained from ALooper::GetNowUs()
+ // return the number of us elapsed since Jan 1 1970 00:00:00 (UTC).
+ int64_t adjustTimeUs(int64_t timeUs) const;
+
+private:
+ enum {
+ kNTPPort = 123,
+ kNTPPacketSize = 48,
+ kNTPModeClient = 3,
+ kNTPVersion = 3,
+ kNTPTransmitTimeOffset = 40,
+ kNTPOriginateTimeOffset = 24,
+ kNTPReceiveTimeOffset = 32,
+ };
+
+ uint64_t mTimeReferenceNTP;
+ int64_t mTimeReferenceUs;
+ int64_t mRoundTripTimeNTP;
+
+ static void writeTimeStamp(uint8_t *dst, uint64_t ntpTime);
+ static uint64_t readTimeStamp(const uint8_t *dst);
+
+ static uint64_t getNowNTP();
+ static uint64_t makeNTP(uint64_t deltaUs);
+
+ DISALLOW_EVIL_CONSTRUCTORS(SNTPClient);
+};
+
+} // namespace android
+
+#endif // SNTP_CLIENT_H_
diff --git a/media/libstagefright/wifi-display/TimeSyncer.cpp b/media/libstagefright/wifi-display/TimeSyncer.cpp
new file mode 100644
index 0000000..cb429bc
--- /dev/null
+++ b/media/libstagefright/wifi-display/TimeSyncer.cpp
@@ -0,0 +1,338 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NEBUG 0
+#define LOG_TAG "TimeSyncer"
+#include <utils/Log.h>
+
+#include "TimeSyncer.h"
+
+#include "ANetworkSession.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+TimeSyncer::TimeSyncer(
+ const sp<ANetworkSession> &netSession, const sp<AMessage> ¬ify)
+ : mNetSession(netSession),
+ mNotify(notify),
+ mIsServer(false),
+ mConnected(false),
+ mUDPSession(0),
+ mSeqNo(0),
+ mTotalTimeUs(0.0),
+ mPendingT1(0ll),
+ mTimeoutGeneration(0) {
+}
+
+TimeSyncer::~TimeSyncer() {
+}
+
+void TimeSyncer::startServer(unsigned localPort) {
+ sp<AMessage> msg = new AMessage(kWhatStartServer, id());
+ msg->setInt32("localPort", localPort);
+ msg->post();
+}
+
+void TimeSyncer::startClient(const char *remoteHost, unsigned remotePort) {
+ sp<AMessage> msg = new AMessage(kWhatStartClient, id());
+ msg->setString("remoteHost", remoteHost);
+ msg->setInt32("remotePort", remotePort);
+ msg->post();
+}
+
+void TimeSyncer::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatStartClient:
+ {
+ AString remoteHost;
+ CHECK(msg->findString("remoteHost", &remoteHost));
+
+ int32_t remotePort;
+ CHECK(msg->findInt32("remotePort", &remotePort));
+
+ sp<AMessage> notify = new AMessage(kWhatUDPNotify, id());
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->createUDPSession(
+ 0 /* localPort */,
+ remoteHost.c_str(),
+ remotePort,
+ notify,
+ &mUDPSession));
+
+ postSendPacket();
+ break;
+ }
+
+ case kWhatStartServer:
+ {
+ mIsServer = true;
+
+ int32_t localPort;
+ CHECK(msg->findInt32("localPort", &localPort));
+
+ sp<AMessage> notify = new AMessage(kWhatUDPNotify, id());
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->createUDPSession(
+ localPort, notify, &mUDPSession));
+
+ break;
+ }
+
+ case kWhatSendPacket:
+ {
+ if (mHistory.size() == 0) {
+ ALOGI("starting batch");
+ }
+
+ TimeInfo ti;
+ memset(&ti, 0, sizeof(ti));
+
+ ti.mT1 = ALooper::GetNowUs();
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->sendRequest(
+ mUDPSession, &ti, sizeof(ti)));
+
+ mPendingT1 = ti.mT1;
+ postTimeout();
+ break;
+ }
+
+ case kWhatTimedOut:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mTimeoutGeneration) {
+ break;
+ }
+
+ ALOGI("timed out, sending another request");
+ postSendPacket();
+ break;
+ }
+
+ case kWhatUDPNotify:
+ {
+ int32_t reason;
+ CHECK(msg->findInt32("reason", &reason));
+
+ switch (reason) {
+ case ANetworkSession::kWhatError:
+ {
+ int32_t sessionID;
+ CHECK(msg->findInt32("sessionID", &sessionID));
+
+ int32_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ AString detail;
+ CHECK(msg->findString("detail", &detail));
+
+ ALOGE("An error occurred in session %d (%d, '%s/%s').",
+ sessionID,
+ err,
+ detail.c_str(),
+ strerror(-err));
+
+ mNetSession->destroySession(sessionID);
+
+ cancelTimeout();
+
+ notifyError(err);
+ break;
+ }
+
+ case ANetworkSession::kWhatDatagram:
+ {
+ int32_t sessionID;
+ CHECK(msg->findInt32("sessionID", &sessionID));
+
+ sp<ABuffer> packet;
+ CHECK(msg->findBuffer("data", &packet));
+
+ int64_t arrivalTimeUs;
+ CHECK(packet->meta()->findInt64(
+ "arrivalTimeUs", &arrivalTimeUs));
+
+ CHECK_EQ(packet->size(), sizeof(TimeInfo));
+
+ TimeInfo *ti = (TimeInfo *)packet->data();
+
+ if (mIsServer) {
+ if (!mConnected) {
+ AString fromAddr;
+ CHECK(msg->findString("fromAddr", &fromAddr));
+
+ int32_t fromPort;
+ CHECK(msg->findInt32("fromPort", &fromPort));
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->connectUDPSession(
+ mUDPSession, fromAddr.c_str(), fromPort));
+
+ mConnected = true;
+ }
+
+ ti->mT2 = arrivalTimeUs;
+ ti->mT3 = ALooper::GetNowUs();
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->sendRequest(
+ mUDPSession, ti, sizeof(*ti)));
+ } else {
+ if (ti->mT1 != mPendingT1) {
+ break;
+ }
+
+ cancelTimeout();
+ mPendingT1 = 0;
+
+ ti->mT4 = arrivalTimeUs;
+
+ // One way delay for a packet to travel from client
+ // to server or back (assumed to be the same either way).
+ int64_t delay =
+ (ti->mT2 - ti->mT1 + ti->mT4 - ti->mT3) / 2;
+
+ // Offset between the client clock (T1, T4) and the
+ // server clock (T2, T3) timestamps.
+ int64_t offset =
+ (ti->mT2 - ti->mT1 - ti->mT4 + ti->mT3) / 2;
+
+ mHistory.push_back(*ti);
+
+ ALOGV("delay = %lld us,\toffset %lld us",
+ delay,
+ offset);
+
+ if (mHistory.size() < kNumPacketsPerBatch) {
+ postSendPacket(1000000ll / 30);
+ } else {
+ notifyOffset();
+
+ ALOGI("batch done");
+
+ mHistory.clear();
+ postSendPacket(kBatchDelayUs);
+ }
+ }
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void TimeSyncer::postSendPacket(int64_t delayUs) {
+ (new AMessage(kWhatSendPacket, id()))->post(delayUs);
+}
+
+void TimeSyncer::postTimeout() {
+ sp<AMessage> msg = new AMessage(kWhatTimedOut, id());
+ msg->setInt32("generation", mTimeoutGeneration);
+ msg->post(kTimeoutDelayUs);
+}
+
+void TimeSyncer::cancelTimeout() {
+ ++mTimeoutGeneration;
+}
+
+void TimeSyncer::notifyError(status_t err) {
+ if (mNotify == NULL) {
+ looper()->stop();
+ return;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatError);
+ notify->setInt32("err", err);
+ notify->post();
+}
+
+// static
+int TimeSyncer::CompareRountripTime(const TimeInfo *ti1, const TimeInfo *ti2) {
+ int64_t rt1 = ti1->mT4 - ti1->mT1;
+ int64_t rt2 = ti2->mT4 - ti2->mT1;
+
+ if (rt1 < rt2) {
+ return -1;
+ } else if (rt1 > rt2) {
+ return 1;
+ }
+
+ return 0;
+}
+
+void TimeSyncer::notifyOffset() {
+ mHistory.sort(CompareRountripTime);
+
+ int64_t sum = 0ll;
+ size_t count = 0;
+
+ // Only consider the third of the information associated with the best
+ // (smallest) roundtrip times.
+ for (size_t i = 0; i < mHistory.size() / 3; ++i) {
+ const TimeInfo *ti = &mHistory[i];
+
+#if 0
+ // One way delay for a packet to travel from client
+ // to server or back (assumed to be the same either way).
+ int64_t delay =
+ (ti->mT2 - ti->mT1 + ti->mT4 - ti->mT3) / 2;
+#endif
+
+ // Offset between the client clock (T1, T4) and the
+ // server clock (T2, T3) timestamps.
+ int64_t offset =
+ (ti->mT2 - ti->mT1 - ti->mT4 + ti->mT3) / 2;
+
+ ALOGV("(%d) RT: %lld us, offset: %lld us",
+ i, ti->mT4 - ti->mT1, offset);
+
+ sum += offset;
+ ++count;
+ }
+
+ if (mNotify == NULL) {
+ ALOGI("avg. offset is %lld", sum / count);
+ return;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatTimeOffset);
+ notify->setInt64("offset", sum / count);
+ notify->post();
+}
+
+} // namespace android
diff --git a/media/libstagefright/wifi-display/TimeSyncer.h b/media/libstagefright/wifi-display/TimeSyncer.h
new file mode 100644
index 0000000..4e7571f
--- /dev/null
+++ b/media/libstagefright/wifi-display/TimeSyncer.h
@@ -0,0 +1,109 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIME_SYNCER_H_
+
+#define TIME_SYNCER_H_
+
+#include <media/stagefright/foundation/AHandler.h>
+
+namespace android {
+
+struct ANetworkSession;
+
+/*
+ TimeSyncer allows us to synchronize time between a client and a server.
+ The client sends a UDP packet containing its send-time to the server,
+ the server sends that packet back to the client amended with information
+ about when it was received as well as the time the reply was sent back.
+ Finally the client receives the reply and has now enough information to
+ compute the clock offset between client and server assuming that packet
+ exchange is symmetric, i.e. time for a packet client->server and
+ server->client is roughly equal.
+ This exchange is repeated a number of times and the average offset computed
+ over the 30% of packets that had the lowest roundtrip times.
+ The offset is determined every 10 secs to account for slight differences in
+ clock frequency.
+*/
+struct TimeSyncer : public AHandler {
+ enum {
+ kWhatError,
+ kWhatTimeOffset,
+ };
+ TimeSyncer(
+ const sp<ANetworkSession> &netSession,
+ const sp<AMessage> ¬ify);
+
+ void startServer(unsigned localPort);
+ void startClient(const char *remoteHost, unsigned remotePort);
+
+protected:
+ virtual ~TimeSyncer();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kWhatStartServer,
+ kWhatStartClient,
+ kWhatUDPNotify,
+ kWhatSendPacket,
+ kWhatTimedOut,
+ };
+
+ struct TimeInfo {
+ int64_t mT1; // client timestamp at send
+ int64_t mT2; // server timestamp at receive
+ int64_t mT3; // server timestamp at send
+ int64_t mT4; // client timestamp at receive
+ };
+
+ enum {
+ kNumPacketsPerBatch = 30,
+ };
+ static const int64_t kTimeoutDelayUs = 500000ll;
+ static const int64_t kBatchDelayUs = 60000000ll; // every minute
+
+ sp<ANetworkSession> mNetSession;
+ sp<AMessage> mNotify;
+
+ bool mIsServer;
+ bool mConnected;
+ int32_t mUDPSession;
+ uint32_t mSeqNo;
+ double mTotalTimeUs;
+
+ Vector<TimeInfo> mHistory;
+
+ int64_t mPendingT1;
+ int32_t mTimeoutGeneration;
+
+ void postSendPacket(int64_t delayUs = 0ll);
+
+ void postTimeout();
+ void cancelTimeout();
+
+ void notifyError(status_t err);
+ void notifyOffset();
+
+ static int CompareRountripTime(const TimeInfo *ti1, const TimeInfo *ti2);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TimeSyncer);
+};
+
+} // namespace android
+
+#endif // TIME_SYNCER_H_
diff --git a/media/libstagefright/wifi-display/VideoFormats.cpp b/media/libstagefright/wifi-display/VideoFormats.cpp
index 458b163..04e02c1 100644
--- a/media/libstagefright/wifi-display/VideoFormats.cpp
+++ b/media/libstagefright/wifi-display/VideoFormats.cpp
@@ -24,7 +24,8 @@
namespace android {
-VideoFormats::config_t VideoFormats::mConfigs[][32] = {
+// static
+const VideoFormats::config_t VideoFormats::mResolutionTable[][32] = {
{
// CEA Resolutions
{ 640, 480, 60, false, 0, 0},
@@ -133,6 +134,8 @@
};
VideoFormats::VideoFormats() {
+ memcpy(mConfigs, mResolutionTable, sizeof(mConfigs));
+
for (size_t i = 0; i < kNumResolutionTypes; ++i) {
mResolutionEnabled[i] = 0;
}
@@ -175,6 +178,29 @@
}
}
+void VideoFormats::enableResolutionUpto(
+ ResolutionType type, size_t index,
+ ProfileType profile, LevelType level) {
+ size_t width, height, fps, score;
+ bool interlaced;
+ if (!GetConfiguration(type, index, &width, &height,
+ &fps, &interlaced)) {
+ ALOGE("Maximum resolution not found!");
+ return;
+ }
+ score = width * height * fps * (!interlaced + 1);
+ for (size_t i = 0; i < kNumResolutionTypes; ++i) {
+ for (size_t j = 0; j < 32; j++) {
+ if (GetConfiguration((ResolutionType)i, j,
+ &width, &height, &fps, &interlaced)
+ && score >= width * height * fps * (!interlaced + 1)) {
+ setResolutionEnabled((ResolutionType)i, j);
+ setProfileLevel((ResolutionType)i, j, profile, level);
+ }
+ }
+ }
+}
+
void VideoFormats::setResolutionEnabled(
ResolutionType type, size_t index, bool enabled) {
CHECK_LT(type, kNumResolutionTypes);
@@ -182,11 +208,56 @@
if (enabled) {
mResolutionEnabled[type] |= (1ul << index);
+ mConfigs[type][index].profile = (1ul << PROFILE_CBP);
+ mConfigs[type][index].level = (1ul << LEVEL_31);
} else {
mResolutionEnabled[type] &= ~(1ul << index);
+ mConfigs[type][index].profile = 0;
+ mConfigs[type][index].level = 0;
}
}
+void VideoFormats::setProfileLevel(
+ ResolutionType type, size_t index,
+ ProfileType profile, LevelType level) {
+ CHECK_LT(type, kNumResolutionTypes);
+ CHECK(GetConfiguration(type, index, NULL, NULL, NULL, NULL));
+
+ mConfigs[type][index].profile = (1ul << profile);
+ mConfigs[type][index].level = (1ul << level);
+}
+
+void VideoFormats::getProfileLevel(
+ ResolutionType type, size_t index,
+ ProfileType *profile, LevelType *level) const{
+ CHECK_LT(type, kNumResolutionTypes);
+ CHECK(GetConfiguration(type, index, NULL, NULL, NULL, NULL));
+
+ int i, bestProfile = -1, bestLevel = -1;
+
+ for (i = 0; i < kNumProfileTypes; ++i) {
+ if (mConfigs[type][index].profile & (1ul << i)) {
+ bestProfile = i;
+ }
+ }
+
+ for (i = 0; i < kNumLevelTypes; ++i) {
+ if (mConfigs[type][index].level & (1ul << i)) {
+ bestLevel = i;
+ }
+ }
+
+ if (bestProfile == -1 || bestLevel == -1) {
+ ALOGE("Profile or level not set for resolution type %d, index %d",
+ type, index);
+ bestProfile = PROFILE_CBP;
+ bestLevel = LEVEL_31;
+ }
+
+ *profile = (ProfileType) bestProfile;
+ *level = (LevelType) bestLevel;
+}
+
bool VideoFormats::isResolutionEnabled(
ResolutionType type, size_t index) const {
CHECK_LT(type, kNumResolutionTypes);
@@ -207,7 +278,7 @@
return false;
}
- const config_t *config = &mConfigs[type][index];
+ const config_t *config = &mResolutionTable[type][index];
if (config->width == 0) {
return false;
@@ -251,9 +322,12 @@
if (res[i] & (1ul << j)){
mResolutionEnabled[i] |= (1ul << j);
if (profile > mConfigs[i][j].profile) {
+ // prefer higher profile (even if level is lower)
mConfigs[i][j].profile = profile;
- if (level > mConfigs[i][j].level)
- mConfigs[i][j].level = level;
+ mConfigs[i][j].level = level;
+ } else if (profile == mConfigs[i][j].profile &&
+ level > mConfigs[i][j].level) {
+ mConfigs[i][j].level = level;
}
}
}
@@ -262,9 +336,51 @@
return true;
}
+// static
+bool VideoFormats::GetProfileLevel(
+ ProfileType profile, LevelType level, unsigned *profileIdc,
+ unsigned *levelIdc, unsigned *constraintSet) {
+ CHECK_LT(profile, kNumProfileTypes);
+ CHECK_LT(level, kNumLevelTypes);
+
+ static const unsigned kProfileIDC[kNumProfileTypes] = {
+ 66, // PROFILE_CBP
+ 100, // PROFILE_CHP
+ };
+
+ static const unsigned kLevelIDC[kNumLevelTypes] = {
+ 31, // LEVEL_31
+ 32, // LEVEL_32
+ 40, // LEVEL_40
+ 41, // LEVEL_41
+ 42, // LEVEL_42
+ };
+
+ static const unsigned kConstraintSet[kNumProfileTypes] = {
+ 0xc0, // PROFILE_CBP
+ 0x0c, // PROFILE_CHP
+ };
+
+ if (profileIdc) {
+ *profileIdc = kProfileIDC[profile];
+ }
+
+ if (levelIdc) {
+ *levelIdc = kLevelIDC[level];
+ }
+
+ if (constraintSet) {
+ *constraintSet = kConstraintSet[profile];
+ }
+
+ return true;
+}
+
bool VideoFormats::parseFormatSpec(const char *spec) {
CHECK_EQ(kNumResolutionTypes, 3);
+ disableAll();
+
unsigned native, dummy;
unsigned res[3];
size_t size = strlen(spec);
@@ -320,8 +436,10 @@
// max-vres (none or 2 byte)
return StringPrintf(
- "%02x 00 02 02 %08x %08x %08x 00 0000 0000 00 none none",
+ "%02x 00 %02x %02x %08x %08x %08x 00 0000 0000 00 none none",
forM4Message ? 0x00 : ((mNativeIndex << 3) | mNativeType),
+ mConfigs[mNativeType][mNativeIndex].profile,
+ mConfigs[mNativeType][mNativeIndex].level,
mResolutionEnabled[0],
mResolutionEnabled[1],
mResolutionEnabled[2]);
@@ -332,7 +450,9 @@
const VideoFormats &sinkSupported,
const VideoFormats &sourceSupported,
ResolutionType *chosenType,
- size_t *chosenIndex) {
+ size_t *chosenIndex,
+ ProfileType *chosenProfile,
+ LevelType *chosenLevel) {
#if 0
// Support for the native format is a great idea, the spec includes
// these features, but nobody supports it and the tests don't validate it.
@@ -412,6 +532,18 @@
*chosenType = (ResolutionType)bestType;
*chosenIndex = bestIndex;
+ // Pick the best profile/level supported by both sink and source.
+ ProfileType srcProfile, sinkProfile;
+ LevelType srcLevel, sinkLevel;
+ sourceSupported.getProfileLevel(
+ (ResolutionType)bestType, bestIndex,
+ &srcProfile, &srcLevel);
+ sinkSupported.getProfileLevel(
+ (ResolutionType)bestType, bestIndex,
+ &sinkProfile, &sinkLevel);
+ *chosenProfile = srcProfile < sinkProfile ? srcProfile : sinkProfile;
+ *chosenLevel = srcLevel < sinkLevel ? srcLevel : sinkLevel;
+
return true;
}
diff --git a/media/libstagefright/wifi-display/VideoFormats.h b/media/libstagefright/wifi-display/VideoFormats.h
index 01de246..fd38fd1 100644
--- a/media/libstagefright/wifi-display/VideoFormats.h
+++ b/media/libstagefright/wifi-display/VideoFormats.h
@@ -69,17 +69,33 @@
void disableAll();
void enableAll();
+ void enableResolutionUpto(
+ ResolutionType type, size_t index,
+ ProfileType profile, LevelType level);
void setResolutionEnabled(
ResolutionType type, size_t index, bool enabled = true);
bool isResolutionEnabled(ResolutionType type, size_t index) const;
+ void setProfileLevel(
+ ResolutionType type, size_t index,
+ ProfileType profile, LevelType level);
+
+ void getProfileLevel(
+ ResolutionType type, size_t index,
+ ProfileType *profile, LevelType *level) const;
+
static bool GetConfiguration(
ResolutionType type, size_t index,
size_t *width, size_t *height, size_t *framesPerSecond,
bool *interlaced);
+ static bool GetProfileLevel(
+ ProfileType profile, LevelType level,
+ unsigned *profileIdc, unsigned *levelIdc,
+ unsigned *constraintSet);
+
bool parseFormatSpec(const char *spec);
AString getFormatSpec(bool forM4Message = false) const;
@@ -87,7 +103,9 @@
const VideoFormats &sinkSupported,
const VideoFormats &sourceSupported,
ResolutionType *chosenType,
- size_t *chosenIndex);
+ size_t *chosenIndex,
+ ProfileType *chosenProfile,
+ LevelType *chosenLevel);
private:
bool parseH264Codec(const char *spec);
@@ -95,7 +113,8 @@
size_t mNativeIndex;
uint32_t mResolutionEnabled[kNumResolutionTypes];
- static config_t mConfigs[kNumResolutionTypes][32];
+ static const config_t mResolutionTable[kNumResolutionTypes][32];
+ config_t mConfigs[kNumResolutionTypes][32];
DISALLOW_EVIL_CONSTRUCTORS(VideoFormats);
};
diff --git a/media/libstagefright/wifi-display/nettest.cpp b/media/libstagefright/wifi-display/nettest.cpp
new file mode 100644
index 0000000..0779bf5
--- /dev/null
+++ b/media/libstagefright/wifi-display/nettest.cpp
@@ -0,0 +1,400 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NEBUG 0
+#define LOG_TAG "nettest"
+#include <utils/Log.h>
+
+#include "ANetworkSession.h"
+#include "TimeSyncer.h"
+
+#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/NuMediaExtractor.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+struct TestHandler : public AHandler {
+ TestHandler(const sp<ANetworkSession> &netSession);
+
+ void listen(int32_t port);
+ void connect(const char *host, int32_t port);
+
+protected:
+ virtual ~TestHandler();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kTimeSyncerPort = 8123,
+ };
+
+ enum {
+ kWhatListen,
+ kWhatConnect,
+ kWhatTimeSyncerNotify,
+ kWhatNetNotify,
+ kWhatSendMore,
+ kWhatStop,
+ };
+
+ sp<ANetworkSession> mNetSession;
+ sp<TimeSyncer> mTimeSyncer;
+
+ int32_t mServerSessionID;
+ int32_t mSessionID;
+
+ int64_t mTimeOffsetUs;
+ bool mTimeOffsetValid;
+
+ int32_t mCounter;
+
+ int64_t mMaxDelayMs;
+
+ void dumpDelay(int32_t counter, int64_t delayMs);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TestHandler);
+};
+
+TestHandler::TestHandler(const sp<ANetworkSession> &netSession)
+ : mNetSession(netSession),
+ mServerSessionID(0),
+ mSessionID(0),
+ mTimeOffsetUs(-1ll),
+ mTimeOffsetValid(false),
+ mCounter(0),
+ mMaxDelayMs(-1ll) {
+}
+
+TestHandler::~TestHandler() {
+}
+
+void TestHandler::listen(int32_t port) {
+ sp<AMessage> msg = new AMessage(kWhatListen, id());
+ msg->setInt32("port", port);
+ msg->post();
+}
+
+void TestHandler::connect(const char *host, int32_t port) {
+ sp<AMessage> msg = new AMessage(kWhatConnect, id());
+ msg->setString("host", host);
+ msg->setInt32("port", port);
+ msg->post();
+}
+
+void TestHandler::dumpDelay(int32_t counter, int64_t delayMs) {
+ static const int64_t kMinDelayMs = 0;
+ static const int64_t kMaxDelayMs = 300;
+
+ const char *kPattern = "########################################";
+ size_t kPatternSize = strlen(kPattern);
+
+ int n = (kPatternSize * (delayMs - kMinDelayMs))
+ / (kMaxDelayMs - kMinDelayMs);
+
+ if (n < 0) {
+ n = 0;
+ } else if ((size_t)n > kPatternSize) {
+ n = kPatternSize;
+ }
+
+ if (delayMs > mMaxDelayMs) {
+ mMaxDelayMs = delayMs;
+ }
+
+ ALOGI("[%d] (%4lld ms / %4lld ms) %s",
+ counter,
+ delayMs,
+ mMaxDelayMs,
+ kPattern + kPatternSize - n);
+}
+
+void TestHandler::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatListen:
+ {
+ sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
+ mTimeSyncer = new TimeSyncer(mNetSession, notify);
+ looper()->registerHandler(mTimeSyncer);
+
+ notify = new AMessage(kWhatNetNotify, id());
+
+ int32_t port;
+ CHECK(msg->findInt32("port", &port));
+
+ struct in_addr ifaceAddr;
+ ifaceAddr.s_addr = INADDR_ANY;
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->createTCPDatagramSession(
+ ifaceAddr,
+ port,
+ notify,
+ &mServerSessionID));
+ break;
+ }
+
+ case kWhatConnect:
+ {
+ sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
+ mTimeSyncer = new TimeSyncer(mNetSession, notify);
+ looper()->registerHandler(mTimeSyncer);
+ mTimeSyncer->startServer(kTimeSyncerPort);
+
+ AString host;
+ CHECK(msg->findString("host", &host));
+
+ int32_t port;
+ CHECK(msg->findInt32("port", &port));
+
+ notify = new AMessage(kWhatNetNotify, id());
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->createTCPDatagramSession(
+ 0 /* localPort */,
+ host.c_str(),
+ port,
+ notify,
+ &mSessionID));
+ break;
+ }
+
+ case kWhatNetNotify:
+ {
+ int32_t reason;
+ CHECK(msg->findInt32("reason", &reason));
+
+ switch (reason) {
+ case ANetworkSession::kWhatConnected:
+ {
+ ALOGI("kWhatConnected");
+
+ (new AMessage(kWhatSendMore, id()))->post();
+ break;
+ }
+
+ case ANetworkSession::kWhatClientConnected:
+ {
+ ALOGI("kWhatClientConnected");
+
+ CHECK_EQ(mSessionID, 0);
+ CHECK(msg->findInt32("sessionID", &mSessionID));
+
+ AString clientIP;
+ CHECK(msg->findString("client-ip", &clientIP));
+
+ mTimeSyncer->startClient(clientIP.c_str(), kTimeSyncerPort);
+ break;
+ }
+
+ case ANetworkSession::kWhatDatagram:
+ {
+ sp<ABuffer> packet;
+ CHECK(msg->findBuffer("data", &packet));
+
+ CHECK_EQ(packet->size(), 12u);
+
+ int32_t counter = U32_AT(packet->data());
+ int64_t timeUs = U64_AT(packet->data() + 4);
+
+ if (mTimeOffsetValid) {
+ timeUs -= mTimeOffsetUs;
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t delayMs = (nowUs - timeUs) / 1000ll;
+
+ dumpDelay(counter, delayMs);
+ } else {
+ ALOGI("received %d", counter);
+ }
+ break;
+ }
+
+ case ANetworkSession::kWhatError:
+ {
+ ALOGE("kWhatError");
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+ break;
+ }
+
+ case kWhatTimeSyncerNotify:
+ {
+ CHECK(msg->findInt64("offset", &mTimeOffsetUs));
+ mTimeOffsetValid = true;
+ break;
+ }
+
+ case kWhatSendMore:
+ {
+ uint8_t buffer[4 + 8];
+ buffer[0] = mCounter >> 24;
+ buffer[1] = (mCounter >> 16) & 0xff;
+ buffer[2] = (mCounter >> 8) & 0xff;
+ buffer[3] = mCounter & 0xff;
+
+ int64_t nowUs = ALooper::GetNowUs();
+
+ buffer[4] = nowUs >> 56;
+ buffer[5] = (nowUs >> 48) & 0xff;
+ buffer[6] = (nowUs >> 40) & 0xff;
+ buffer[7] = (nowUs >> 32) & 0xff;
+ buffer[8] = (nowUs >> 24) & 0xff;
+ buffer[9] = (nowUs >> 16) & 0xff;
+ buffer[10] = (nowUs >> 8) & 0xff;
+ buffer[11] = nowUs & 0xff;
+
+ ++mCounter;
+
+ CHECK_EQ((status_t)OK,
+ mNetSession->sendRequest(
+ mSessionID,
+ buffer,
+ sizeof(buffer),
+ true /* timeValid */,
+ nowUs));
+
+ msg->post(100000ll);
+ break;
+ }
+
+ case kWhatStop:
+ {
+ if (mSessionID != 0) {
+ mNetSession->destroySession(mSessionID);
+ mSessionID = 0;
+ }
+
+ if (mServerSessionID != 0) {
+ mNetSession->destroySession(mServerSessionID);
+ mServerSessionID = 0;
+ }
+
+ looper()->stop();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+} // namespace android
+
+static void usage(const char *me) {
+ fprintf(stderr,
+ "usage: %s -c host:port\tconnect to remote host\n"
+ " -l port \tlisten\n",
+ me);
+}
+
+int main(int argc, char **argv) {
+ using namespace android;
+
+ // srand(time(NULL));
+
+ ProcessState::self()->startThreadPool();
+
+ DataSource::RegisterDefaultSniffers();
+
+ int32_t connectToPort = -1;
+ AString connectToHost;
+
+ int32_t listenOnPort = -1;
+
+ int res;
+ while ((res = getopt(argc, argv, "hc:l:")) >= 0) {
+ switch (res) {
+ case 'c':
+ {
+ const char *colonPos = strrchr(optarg, ':');
+
+ if (colonPos == NULL) {
+ usage(argv[0]);
+ exit(1);
+ }
+
+ connectToHost.setTo(optarg, colonPos - optarg);
+
+ char *end;
+ connectToPort = strtol(colonPos + 1, &end, 10);
+
+ if (*end != '\0' || end == colonPos + 1
+ || connectToPort < 0 || connectToPort > 65535) {
+ fprintf(stderr, "Illegal port specified.\n");
+ exit(1);
+ }
+ break;
+ }
+
+ case 'l':
+ {
+ char *end;
+ listenOnPort = strtol(optarg, &end, 10);
+
+ if (*end != '\0' || end == optarg
+ || listenOnPort < 0 || listenOnPort > 65535) {
+ fprintf(stderr, "Illegal port specified.\n");
+ exit(1);
+ }
+ break;
+ }
+
+ case '?':
+ case 'h':
+ usage(argv[0]);
+ exit(1);
+ }
+ }
+
+ if ((listenOnPort < 0 && connectToPort < 0)
+ || (listenOnPort >= 0 && connectToPort >= 0)) {
+ fprintf(stderr,
+ "You need to select either client or server mode.\n");
+ exit(1);
+ }
+
+ sp<ANetworkSession> netSession = new ANetworkSession;
+ netSession->start();
+
+ sp<ALooper> looper = new ALooper;
+
+ sp<TestHandler> handler = new TestHandler(netSession);
+ looper->registerHandler(handler);
+
+ if (listenOnPort) {
+ handler->listen(listenOnPort);
+ }
+
+ if (connectToPort >= 0) {
+ handler->connect(connectToHost.c_str(), connectToPort);
+ }
+
+ looper->start(true /* runOnCallingThread */);
+
+ return 0;
+}
diff --git a/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp b/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp
new file mode 100644
index 0000000..7a96081
--- /dev/null
+++ b/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp
@@ -0,0 +1,328 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "RTPAssembler"
+#include <utils/Log.h>
+
+#include "RTPAssembler.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/MediaErrors.h>
+
+namespace android {
+
+RTPReceiver::Assembler::Assembler(const sp<AMessage> ¬ify)
+ : mNotify(notify) {
+}
+
+void RTPReceiver::Assembler::postAccessUnit(
+ const sp<ABuffer> &accessUnit, bool followsDiscontinuity) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", RTPReceiver::kWhatAccessUnit);
+ notify->setBuffer("accessUnit", accessUnit);
+ notify->setInt32("followsDiscontinuity", followsDiscontinuity);
+ notify->post();
+}
+////////////////////////////////////////////////////////////////////////////////
+
+RTPReceiver::TSAssembler::TSAssembler(const sp<AMessage> ¬ify)
+ : Assembler(notify),
+ mSawDiscontinuity(false) {
+}
+
+void RTPReceiver::TSAssembler::signalDiscontinuity() {
+ mSawDiscontinuity = true;
+}
+
+status_t RTPReceiver::TSAssembler::processPacket(const sp<ABuffer> &packet) {
+ int32_t rtpTime;
+ CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
+
+ packet->meta()->setInt64("timeUs", (rtpTime * 100ll) / 9);
+
+ postAccessUnit(packet, mSawDiscontinuity);
+
+ if (mSawDiscontinuity) {
+ mSawDiscontinuity = false;
+ }
+
+ return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+RTPReceiver::H264Assembler::H264Assembler(const sp<AMessage> ¬ify)
+ : Assembler(notify),
+ mState(0),
+ mIndicator(0),
+ mNALType(0),
+ mAccessUnitRTPTime(0) {
+}
+
+void RTPReceiver::H264Assembler::signalDiscontinuity() {
+ reset();
+}
+
+status_t RTPReceiver::H264Assembler::processPacket(const sp<ABuffer> &packet) {
+ status_t err = internalProcessPacket(packet);
+
+ if (err != OK) {
+ reset();
+ }
+
+ return err;
+}
+
+status_t RTPReceiver::H264Assembler::internalProcessPacket(
+ const sp<ABuffer> &packet) {
+ const uint8_t *data = packet->data();
+ size_t size = packet->size();
+
+ switch (mState) {
+ case 0:
+ {
+ if (size < 1 || (data[0] & 0x80)) {
+ ALOGV("Malformed H264 RTP packet (empty or F-bit set)");
+ return ERROR_MALFORMED;
+ }
+
+ unsigned nalType = data[0] & 0x1f;
+ if (nalType >= 1 && nalType <= 23) {
+ addSingleNALUnit(packet);
+ ALOGV("added single NAL packet");
+ } else if (nalType == 28) {
+ // FU-A
+ unsigned indicator = data[0];
+ CHECK((indicator & 0x1f) == 28);
+
+ if (size < 2) {
+ ALOGV("Malformed H264 FU-A packet (single byte)");
+ return ERROR_MALFORMED;
+ }
+
+ if (!(data[1] & 0x80)) {
+ ALOGV("Malformed H264 FU-A packet (no start bit)");
+ return ERROR_MALFORMED;
+ }
+
+ mIndicator = data[0];
+ mNALType = data[1] & 0x1f;
+ uint32_t nri = (data[0] >> 5) & 3;
+
+ clearAccumulator();
+
+ uint8_t byte = mNALType | (nri << 5);
+ appendToAccumulator(&byte, 1);
+ appendToAccumulator(data + 2, size - 2);
+
+ int32_t rtpTime;
+ CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
+ mAccumulator->meta()->setInt32("rtp-time", rtpTime);
+
+ if (data[1] & 0x40) {
+ // Huh? End bit also set on the first buffer.
+ addSingleNALUnit(mAccumulator);
+ clearAccumulator();
+
+ ALOGV("added FU-A");
+ break;
+ }
+
+ mState = 1;
+ } else if (nalType == 24) {
+ // STAP-A
+
+ status_t err = addSingleTimeAggregationPacket(packet);
+ if (err != OK) {
+ return err;
+ }
+ } else {
+ ALOGV("Malformed H264 packet (unknown type %d)", nalType);
+ return ERROR_UNSUPPORTED;
+ }
+ break;
+ }
+
+ case 1:
+ {
+ if (size < 2
+ || data[0] != mIndicator
+ || (data[1] & 0x1f) != mNALType
+ || (data[1] & 0x80)) {
+ ALOGV("Malformed H264 FU-A packet (indicator, "
+ "type or start bit mismatch)");
+
+ return ERROR_MALFORMED;
+ }
+
+ appendToAccumulator(data + 2, size - 2);
+
+ if (data[1] & 0x40) {
+ addSingleNALUnit(mAccumulator);
+
+ clearAccumulator();
+ mState = 0;
+
+ ALOGV("added FU-A");
+ }
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+
+ int32_t marker;
+ CHECK(packet->meta()->findInt32("M", &marker));
+
+ if (marker) {
+ flushAccessUnit();
+ }
+
+ return OK;
+}
+
+void RTPReceiver::H264Assembler::reset() {
+ mNALUnits.clear();
+
+ clearAccumulator();
+ mState = 0;
+}
+
+void RTPReceiver::H264Assembler::clearAccumulator() {
+ if (mAccumulator != NULL) {
+ // XXX Too expensive.
+ mAccumulator.clear();
+ }
+}
+
+void RTPReceiver::H264Assembler::appendToAccumulator(
+ const void *data, size_t size) {
+ if (mAccumulator == NULL) {
+ mAccumulator = new ABuffer(size);
+ memcpy(mAccumulator->data(), data, size);
+ return;
+ }
+
+ if (mAccumulator->size() + size > mAccumulator->capacity()) {
+ sp<ABuffer> buf = new ABuffer(mAccumulator->size() + size);
+ memcpy(buf->data(), mAccumulator->data(), mAccumulator->size());
+ buf->setRange(0, mAccumulator->size());
+
+ int32_t rtpTime;
+ if (mAccumulator->meta()->findInt32("rtp-time", &rtpTime)) {
+ buf->meta()->setInt32("rtp-time", rtpTime);
+ }
+
+ mAccumulator = buf;
+ }
+
+ memcpy(mAccumulator->data() + mAccumulator->size(), data, size);
+ mAccumulator->setRange(0, mAccumulator->size() + size);
+}
+
+void RTPReceiver::H264Assembler::addSingleNALUnit(const sp<ABuffer> &packet) {
+ if (mNALUnits.empty()) {
+ int32_t rtpTime;
+ CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
+
+ mAccessUnitRTPTime = rtpTime;
+ }
+
+ mNALUnits.push_back(packet);
+}
+
+void RTPReceiver::H264Assembler::flushAccessUnit() {
+ if (mNALUnits.empty()) {
+ return;
+ }
+
+ size_t totalSize = 0;
+ for (List<sp<ABuffer> >::iterator it = mNALUnits.begin();
+ it != mNALUnits.end(); ++it) {
+ totalSize += 4 + (*it)->size();
+ }
+
+ sp<ABuffer> accessUnit = new ABuffer(totalSize);
+ size_t offset = 0;
+ for (List<sp<ABuffer> >::iterator it = mNALUnits.begin();
+ it != mNALUnits.end(); ++it) {
+ const sp<ABuffer> nalUnit = *it;
+
+ memcpy(accessUnit->data() + offset, "\x00\x00\x00\x01", 4);
+
+ memcpy(accessUnit->data() + offset + 4,
+ nalUnit->data(),
+ nalUnit->size());
+
+ offset += 4 + nalUnit->size();
+ }
+
+ mNALUnits.clear();
+
+ accessUnit->meta()->setInt64("timeUs", mAccessUnitRTPTime * 100ll / 9ll);
+ postAccessUnit(accessUnit, false /* followsDiscontinuity */);
+}
+
+status_t RTPReceiver::H264Assembler::addSingleTimeAggregationPacket(
+ const sp<ABuffer> &packet) {
+ const uint8_t *data = packet->data();
+ size_t size = packet->size();
+
+ if (size < 3) {
+ ALOGV("Malformed H264 STAP-A packet (too small)");
+ return ERROR_MALFORMED;
+ }
+
+ int32_t rtpTime;
+ CHECK(packet->meta()->findInt32("rtp-time", &rtpTime));
+
+ ++data;
+ --size;
+ while (size >= 2) {
+ size_t nalSize = (data[0] << 8) | data[1];
+
+ if (size < nalSize + 2) {
+ ALOGV("Malformed H264 STAP-A packet (incomplete NAL unit)");
+ return ERROR_MALFORMED;
+ }
+
+ sp<ABuffer> unit = new ABuffer(nalSize);
+ memcpy(unit->data(), &data[2], nalSize);
+
+ unit->meta()->setInt32("rtp-time", rtpTime);
+
+ addSingleNALUnit(unit);
+
+ data += 2 + nalSize;
+ size -= 2 + nalSize;
+ }
+
+ if (size != 0) {
+ ALOGV("Unexpected padding at end of STAP-A packet.");
+ }
+
+ ALOGV("added STAP-A");
+
+ return OK;
+}
+
+} // namespace android
+
diff --git a/media/libstagefright/wifi-display/rtp/RTPAssembler.h b/media/libstagefright/wifi-display/rtp/RTPAssembler.h
new file mode 100644
index 0000000..e456d32
--- /dev/null
+++ b/media/libstagefright/wifi-display/rtp/RTPAssembler.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef RTP_ASSEMBLER_H_
+
+#define RTP_ASSEMBLER_H_
+
+#include "RTPReceiver.h"
+
+namespace android {
+
+// A helper class to reassemble the payload of RTP packets into access
+// units depending on the packetization scheme.
+struct RTPReceiver::Assembler : public RefBase {
+ Assembler(const sp<AMessage> ¬ify);
+
+ virtual void signalDiscontinuity() = 0;
+ virtual status_t processPacket(const sp<ABuffer> &packet) = 0;
+
+protected:
+ virtual ~Assembler() {}
+
+ void postAccessUnit(
+ const sp<ABuffer> &accessUnit, bool followsDiscontinuity);
+
+private:
+ sp<AMessage> mNotify;
+
+ DISALLOW_EVIL_CONSTRUCTORS(Assembler);
+};
+
+struct RTPReceiver::TSAssembler : public RTPReceiver::Assembler {
+ TSAssembler(const sp<AMessage> ¬ify);
+
+ virtual void signalDiscontinuity();
+ virtual status_t processPacket(const sp<ABuffer> &packet);
+
+private:
+ bool mSawDiscontinuity;
+
+ DISALLOW_EVIL_CONSTRUCTORS(TSAssembler);
+};
+
+struct RTPReceiver::H264Assembler : public RTPReceiver::Assembler {
+ H264Assembler(const sp<AMessage> ¬ify);
+
+ virtual void signalDiscontinuity();
+ virtual status_t processPacket(const sp<ABuffer> &packet);
+
+private:
+ int32_t mState;
+
+ uint8_t mIndicator;
+ uint8_t mNALType;
+
+ sp<ABuffer> mAccumulator;
+
+ List<sp<ABuffer> > mNALUnits;
+ int32_t mAccessUnitRTPTime;
+
+ status_t internalProcessPacket(const sp<ABuffer> &packet);
+
+ void addSingleNALUnit(const sp<ABuffer> &packet);
+ status_t addSingleTimeAggregationPacket(const sp<ABuffer> &packet);
+
+ void flushAccessUnit();
+
+ void clearAccumulator();
+ void appendToAccumulator(const void *data, size_t size);
+
+ void reset();
+
+ DISALLOW_EVIL_CONSTRUCTORS(H264Assembler);
+};
+
+} // namespace android
+
+#endif // RTP_ASSEMBLER_H_
+
diff --git a/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp b/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp
new file mode 100644
index 0000000..2d22e79
--- /dev/null
+++ b/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp
@@ -0,0 +1,1153 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "RTPReceiver"
+#include <utils/Log.h>
+
+#include "RTPAssembler.h"
+#include "RTPReceiver.h"
+
+#include "ANetworkSession.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/Utils.h>
+
+#define TRACK_PACKET_LOSS 0
+
+namespace android {
+
+////////////////////////////////////////////////////////////////////////////////
+
+struct RTPReceiver::Source : public AHandler {
+ Source(RTPReceiver *receiver, uint32_t ssrc);
+
+ void onPacketReceived(uint16_t seq, const sp<ABuffer> &buffer);
+
+ void addReportBlock(uint32_t ssrc, const sp<ABuffer> &buf);
+
+protected:
+ virtual ~Source();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kWhatRetransmit,
+ kWhatDeclareLost,
+ };
+
+ static const uint32_t kMinSequential = 2;
+ static const uint32_t kMaxDropout = 3000;
+ static const uint32_t kMaxMisorder = 100;
+ static const uint32_t kRTPSeqMod = 1u << 16;
+ static const int64_t kReportIntervalUs = 10000000ll;
+
+ RTPReceiver *mReceiver;
+ uint32_t mSSRC;
+ bool mFirst;
+ uint16_t mMaxSeq;
+ uint32_t mCycles;
+ uint32_t mBaseSeq;
+ uint32_t mReceived;
+ uint32_t mExpectedPrior;
+ uint32_t mReceivedPrior;
+
+ int64_t mFirstArrivalTimeUs;
+ int64_t mFirstRTPTimeUs;
+
+ // Ordered by extended seq number.
+ List<sp<ABuffer> > mPackets;
+
+ enum StatusBits {
+ STATUS_DECLARED_LOST = 1,
+ STATUS_REQUESTED_RETRANSMISSION = 2,
+ STATUS_ARRIVED_LATE = 4,
+ };
+#if TRACK_PACKET_LOSS
+ KeyedVector<int32_t, uint32_t> mLostPackets;
+#endif
+
+ void modifyPacketStatus(int32_t extSeqNo, uint32_t mask);
+
+ int32_t mAwaitingExtSeqNo;
+ bool mRequestedRetransmission;
+
+ int32_t mActivePacketType;
+ sp<Assembler> mActiveAssembler;
+
+ int64_t mNextReportTimeUs;
+
+ int32_t mNumDeclaredLost;
+ int32_t mNumDeclaredLostPrior;
+
+ int32_t mRetransmitGeneration;
+ int32_t mDeclareLostGeneration;
+ bool mDeclareLostTimerPending;
+
+ void queuePacket(const sp<ABuffer> &packet);
+ void dequeueMore();
+
+ sp<ABuffer> getNextPacket();
+ void resync();
+
+ void postRetransmitTimer(int64_t delayUs);
+ void postDeclareLostTimer(int64_t delayUs);
+ void cancelTimers();
+
+ DISALLOW_EVIL_CONSTRUCTORS(Source);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+RTPReceiver::Source::Source(RTPReceiver *receiver, uint32_t ssrc)
+ : mReceiver(receiver),
+ mSSRC(ssrc),
+ mFirst(true),
+ mMaxSeq(0),
+ mCycles(0),
+ mBaseSeq(0),
+ mReceived(0),
+ mExpectedPrior(0),
+ mReceivedPrior(0),
+ mFirstArrivalTimeUs(-1ll),
+ mFirstRTPTimeUs(-1ll),
+ mAwaitingExtSeqNo(-1),
+ mRequestedRetransmission(false),
+ mActivePacketType(-1),
+ mNextReportTimeUs(-1ll),
+ mNumDeclaredLost(0),
+ mNumDeclaredLostPrior(0),
+ mRetransmitGeneration(0),
+ mDeclareLostGeneration(0),
+ mDeclareLostTimerPending(false) {
+}
+
+RTPReceiver::Source::~Source() {
+}
+
+void RTPReceiver::Source::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatRetransmit:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mRetransmitGeneration) {
+ break;
+ }
+
+ mRequestedRetransmission = true;
+ mReceiver->requestRetransmission(mSSRC, mAwaitingExtSeqNo);
+
+ modifyPacketStatus(
+ mAwaitingExtSeqNo, STATUS_REQUESTED_RETRANSMISSION);
+ break;
+ }
+
+ case kWhatDeclareLost:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mDeclareLostGeneration) {
+ break;
+ }
+
+ cancelTimers();
+
+ ALOGV("Lost packet extSeqNo %d %s",
+ mAwaitingExtSeqNo,
+ mRequestedRetransmission ? "*" : "");
+
+ mRequestedRetransmission = false;
+ if (mActiveAssembler != NULL) {
+ mActiveAssembler->signalDiscontinuity();
+ }
+
+ modifyPacketStatus(mAwaitingExtSeqNo, STATUS_DECLARED_LOST);
+
+ // resync();
+ ++mAwaitingExtSeqNo;
+ ++mNumDeclaredLost;
+
+ mReceiver->notifyPacketLost();
+
+ dequeueMore();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void RTPReceiver::Source::onPacketReceived(
+ uint16_t seq, const sp<ABuffer> &buffer) {
+ if (mFirst) {
+ buffer->setInt32Data(mCycles | seq);
+ queuePacket(buffer);
+
+ mFirst = false;
+ mBaseSeq = seq;
+ mMaxSeq = seq;
+ ++mReceived;
+ return;
+ }
+
+ uint16_t udelta = seq - mMaxSeq;
+
+ if (udelta < kMaxDropout) {
+ // In order, with permissible gap.
+
+ if (seq < mMaxSeq) {
+ // Sequence number wrapped - count another 64K cycle
+ mCycles += kRTPSeqMod;
+ }
+
+ mMaxSeq = seq;
+
+ ++mReceived;
+ } else if (udelta <= kRTPSeqMod - kMaxMisorder) {
+ // The sequence number made a very large jump
+ return;
+ } else {
+ // Duplicate or reordered packet.
+ }
+
+ buffer->setInt32Data(mCycles | seq);
+ queuePacket(buffer);
+}
+
+void RTPReceiver::Source::queuePacket(const sp<ABuffer> &packet) {
+ int32_t newExtendedSeqNo = packet->int32Data();
+
+ if (mFirstArrivalTimeUs < 0ll) {
+ mFirstArrivalTimeUs = ALooper::GetNowUs();
+
+ uint32_t rtpTime;
+ CHECK(packet->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+
+ mFirstRTPTimeUs = (rtpTime * 100ll) / 9ll;
+ }
+
+ if (mAwaitingExtSeqNo >= 0 && newExtendedSeqNo < mAwaitingExtSeqNo) {
+ // We're no longer interested in these. They're old.
+ ALOGV("dropping stale extSeqNo %d", newExtendedSeqNo);
+
+ modifyPacketStatus(newExtendedSeqNo, STATUS_ARRIVED_LATE);
+ return;
+ }
+
+ if (mPackets.empty()) {
+ mPackets.push_back(packet);
+ dequeueMore();
+ return;
+ }
+
+ List<sp<ABuffer> >::iterator firstIt = mPackets.begin();
+ List<sp<ABuffer> >::iterator it = --mPackets.end();
+ for (;;) {
+ int32_t extendedSeqNo = (*it)->int32Data();
+
+ if (extendedSeqNo == newExtendedSeqNo) {
+ // Duplicate packet.
+ return;
+ }
+
+ if (extendedSeqNo < newExtendedSeqNo) {
+ // Insert new packet after the one at "it".
+ mPackets.insert(++it, packet);
+ break;
+ }
+
+ if (it == firstIt) {
+ // Insert new packet before the first existing one.
+ mPackets.insert(it, packet);
+ break;
+ }
+
+ --it;
+ }
+
+ dequeueMore();
+}
+
+void RTPReceiver::Source::dequeueMore() {
+ int64_t nowUs = ALooper::GetNowUs();
+ if (mNextReportTimeUs < 0ll || nowUs >= mNextReportTimeUs) {
+ if (mNextReportTimeUs >= 0ll) {
+ uint32_t expected = (mMaxSeq | mCycles) - mBaseSeq + 1;
+
+ uint32_t expectedInterval = expected - mExpectedPrior;
+ mExpectedPrior = expected;
+
+ uint32_t receivedInterval = mReceived - mReceivedPrior;
+ mReceivedPrior = mReceived;
+
+ int64_t lostInterval =
+ (int64_t)expectedInterval - (int64_t)receivedInterval;
+
+ int32_t declaredLostInterval =
+ mNumDeclaredLost - mNumDeclaredLostPrior;
+
+ mNumDeclaredLostPrior = mNumDeclaredLost;
+
+ if (declaredLostInterval > 0) {
+ ALOGI("lost %lld packets (%.2f %%), declared %d lost\n",
+ lostInterval,
+ 100.0f * lostInterval / expectedInterval,
+ declaredLostInterval);
+ }
+ }
+
+ mNextReportTimeUs = nowUs + kReportIntervalUs;
+
+#if TRACK_PACKET_LOSS
+ for (size_t i = 0; i < mLostPackets.size(); ++i) {
+ int32_t key = mLostPackets.keyAt(i);
+ uint32_t value = mLostPackets.valueAt(i);
+
+ AString status;
+ if (value & STATUS_REQUESTED_RETRANSMISSION) {
+ status.append("retrans ");
+ }
+ if (value & STATUS_ARRIVED_LATE) {
+ status.append("arrived-late ");
+ }
+ ALOGI("Packet %d declared lost %s", key, status.c_str());
+ }
+#endif
+ }
+
+ sp<ABuffer> packet;
+ while ((packet = getNextPacket()) != NULL) {
+ if (mDeclareLostTimerPending) {
+ cancelTimers();
+ }
+
+ CHECK_GE(mAwaitingExtSeqNo, 0);
+#if TRACK_PACKET_LOSS
+ mLostPackets.removeItem(mAwaitingExtSeqNo);
+#endif
+
+ int32_t packetType;
+ CHECK(packet->meta()->findInt32("PT", &packetType));
+
+ if (packetType != mActivePacketType) {
+ mActiveAssembler = mReceiver->makeAssembler(packetType);
+ mActivePacketType = packetType;
+ }
+
+ if (mActiveAssembler != NULL) {
+ status_t err = mActiveAssembler->processPacket(packet);
+ if (err != OK) {
+ ALOGV("assembler returned error %d", err);
+ }
+ }
+
+ ++mAwaitingExtSeqNo;
+ }
+
+ if (mDeclareLostTimerPending) {
+ return;
+ }
+
+ if (mPackets.empty()) {
+ return;
+ }
+
+ CHECK_GE(mAwaitingExtSeqNo, 0);
+
+ const sp<ABuffer> &firstPacket = *mPackets.begin();
+
+ uint32_t rtpTime;
+ CHECK(firstPacket->meta()->findInt32(
+ "rtp-time", (int32_t *)&rtpTime));
+
+
+ int64_t rtpUs = (rtpTime * 100ll) / 9ll;
+
+ int64_t maxArrivalTimeUs =
+ mFirstArrivalTimeUs + rtpUs - mFirstRTPTimeUs;
+
+ nowUs = ALooper::GetNowUs();
+
+ CHECK_LT(mAwaitingExtSeqNo, firstPacket->int32Data());
+
+ ALOGV("waiting for %d, comparing against %d, %lld us left",
+ mAwaitingExtSeqNo,
+ firstPacket->int32Data(),
+ maxArrivalTimeUs - nowUs);
+
+ postDeclareLostTimer(maxArrivalTimeUs + kPacketLostAfterUs);
+
+ if (kRequestRetransmissionAfterUs > 0ll) {
+ postRetransmitTimer(
+ maxArrivalTimeUs + kRequestRetransmissionAfterUs);
+ }
+}
+
+sp<ABuffer> RTPReceiver::Source::getNextPacket() {
+ if (mPackets.empty()) {
+ return NULL;
+ }
+
+ int32_t extSeqNo = (*mPackets.begin())->int32Data();
+
+ if (mAwaitingExtSeqNo < 0) {
+ mAwaitingExtSeqNo = extSeqNo;
+ } else if (extSeqNo != mAwaitingExtSeqNo) {
+ return NULL;
+ }
+
+ sp<ABuffer> packet = *mPackets.begin();
+ mPackets.erase(mPackets.begin());
+
+ return packet;
+}
+
+void RTPReceiver::Source::resync() {
+ mAwaitingExtSeqNo = -1;
+}
+
+void RTPReceiver::Source::addReportBlock(
+ uint32_t ssrc, const sp<ABuffer> &buf) {
+ uint32_t extMaxSeq = mMaxSeq | mCycles;
+ uint32_t expected = extMaxSeq - mBaseSeq + 1;
+
+ int64_t lost = (int64_t)expected - (int64_t)mReceived;
+ if (lost > 0x7fffff) {
+ lost = 0x7fffff;
+ } else if (lost < -0x800000) {
+ lost = -0x800000;
+ }
+
+ uint32_t expectedInterval = expected - mExpectedPrior;
+ mExpectedPrior = expected;
+
+ uint32_t receivedInterval = mReceived - mReceivedPrior;
+ mReceivedPrior = mReceived;
+
+ int64_t lostInterval = expectedInterval - receivedInterval;
+
+ uint8_t fractionLost;
+ if (expectedInterval == 0 || lostInterval <=0) {
+ fractionLost = 0;
+ } else {
+ fractionLost = (lostInterval << 8) / expectedInterval;
+ }
+
+ uint8_t *ptr = buf->data() + buf->size();
+
+ ptr[0] = ssrc >> 24;
+ ptr[1] = (ssrc >> 16) & 0xff;
+ ptr[2] = (ssrc >> 8) & 0xff;
+ ptr[3] = ssrc & 0xff;
+
+ ptr[4] = fractionLost;
+
+ ptr[5] = (lost >> 16) & 0xff;
+ ptr[6] = (lost >> 8) & 0xff;
+ ptr[7] = lost & 0xff;
+
+ ptr[8] = extMaxSeq >> 24;
+ ptr[9] = (extMaxSeq >> 16) & 0xff;
+ ptr[10] = (extMaxSeq >> 8) & 0xff;
+ ptr[11] = extMaxSeq & 0xff;
+
+ // XXX TODO:
+
+ ptr[12] = 0x00; // interarrival jitter
+ ptr[13] = 0x00;
+ ptr[14] = 0x00;
+ ptr[15] = 0x00;
+
+ ptr[16] = 0x00; // last SR
+ ptr[17] = 0x00;
+ ptr[18] = 0x00;
+ ptr[19] = 0x00;
+
+ ptr[20] = 0x00; // delay since last SR
+ ptr[21] = 0x00;
+ ptr[22] = 0x00;
+ ptr[23] = 0x00;
+
+ buf->setRange(buf->offset(), buf->size() + 24);
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+RTPReceiver::RTPReceiver(
+ const sp<ANetworkSession> &netSession,
+ const sp<AMessage> ¬ify,
+ uint32_t flags)
+ : mNetSession(netSession),
+ mNotify(notify),
+ mFlags(flags),
+ mRTPMode(TRANSPORT_UNDEFINED),
+ mRTCPMode(TRANSPORT_UNDEFINED),
+ mRTPSessionID(0),
+ mRTCPSessionID(0),
+ mRTPConnected(false),
+ mRTCPConnected(false),
+ mRTPClientSessionID(0),
+ mRTCPClientSessionID(0) {
+}
+
+RTPReceiver::~RTPReceiver() {
+ if (mRTCPClientSessionID != 0) {
+ mNetSession->destroySession(mRTCPClientSessionID);
+ mRTCPClientSessionID = 0;
+ }
+
+ if (mRTPClientSessionID != 0) {
+ mNetSession->destroySession(mRTPClientSessionID);
+ mRTPClientSessionID = 0;
+ }
+
+ if (mRTCPSessionID != 0) {
+ mNetSession->destroySession(mRTCPSessionID);
+ mRTCPSessionID = 0;
+ }
+
+ if (mRTPSessionID != 0) {
+ mNetSession->destroySession(mRTPSessionID);
+ mRTPSessionID = 0;
+ }
+}
+
+status_t RTPReceiver::initAsync(
+ TransportMode rtpMode,
+ TransportMode rtcpMode,
+ int32_t *outLocalRTPPort) {
+ if (mRTPMode != TRANSPORT_UNDEFINED
+ || rtpMode == TRANSPORT_UNDEFINED
+ || rtpMode == TRANSPORT_NONE
+ || rtcpMode == TRANSPORT_UNDEFINED) {
+ return INVALID_OPERATION;
+ }
+
+ CHECK_NE(rtpMode, TRANSPORT_TCP_INTERLEAVED);
+ CHECK_NE(rtcpMode, TRANSPORT_TCP_INTERLEAVED);
+
+ sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, id());
+
+ sp<AMessage> rtcpNotify;
+ if (rtcpMode != TRANSPORT_NONE) {
+ rtcpNotify = new AMessage(kWhatRTCPNotify, id());
+ }
+
+ CHECK_EQ(mRTPSessionID, 0);
+ CHECK_EQ(mRTCPSessionID, 0);
+
+ int32_t localRTPPort;
+
+ struct in_addr ifaceAddr;
+ ifaceAddr.s_addr = INADDR_ANY;
+
+ for (;;) {
+ localRTPPort = PickRandomRTPPort();
+
+ status_t err;
+ if (rtpMode == TRANSPORT_UDP) {
+ err = mNetSession->createUDPSession(
+ localRTPPort,
+ rtpNotify,
+ &mRTPSessionID);
+ } else {
+ CHECK_EQ(rtpMode, TRANSPORT_TCP);
+ err = mNetSession->createTCPDatagramSession(
+ ifaceAddr,
+ localRTPPort,
+ rtpNotify,
+ &mRTPSessionID);
+ }
+
+ if (err != OK) {
+ continue;
+ }
+
+ if (rtcpMode == TRANSPORT_NONE) {
+ break;
+ } else if (rtcpMode == TRANSPORT_UDP) {
+ err = mNetSession->createUDPSession(
+ localRTPPort + 1,
+ rtcpNotify,
+ &mRTCPSessionID);
+ } else {
+ CHECK_EQ(rtpMode, TRANSPORT_TCP);
+ err = mNetSession->createTCPDatagramSession(
+ ifaceAddr,
+ localRTPPort + 1,
+ rtcpNotify,
+ &mRTCPSessionID);
+ }
+
+ if (err == OK) {
+ break;
+ }
+
+ mNetSession->destroySession(mRTPSessionID);
+ mRTPSessionID = 0;
+ }
+
+ mRTPMode = rtpMode;
+ mRTCPMode = rtcpMode;
+ *outLocalRTPPort = localRTPPort;
+
+ return OK;
+}
+
+status_t RTPReceiver::connect(
+ const char *remoteHost, int32_t remoteRTPPort, int32_t remoteRTCPPort) {
+ status_t err;
+
+ if (mRTPMode == TRANSPORT_UDP) {
+ CHECK(!mRTPConnected);
+
+ err = mNetSession->connectUDPSession(
+ mRTPSessionID, remoteHost, remoteRTPPort);
+
+ if (err != OK) {
+ notifyInitDone(err);
+ return err;
+ }
+
+ ALOGI("connectUDPSession RTP successful.");
+
+ mRTPConnected = true;
+ }
+
+ if (mRTCPMode == TRANSPORT_UDP) {
+ CHECK(!mRTCPConnected);
+
+ err = mNetSession->connectUDPSession(
+ mRTCPSessionID, remoteHost, remoteRTCPPort);
+
+ if (err != OK) {
+ notifyInitDone(err);
+ return err;
+ }
+
+ scheduleSendRR();
+
+ ALOGI("connectUDPSession RTCP successful.");
+
+ mRTCPConnected = true;
+ }
+
+ if (mRTPConnected
+ && (mRTCPConnected || mRTCPMode == TRANSPORT_NONE)) {
+ notifyInitDone(OK);
+ }
+
+ return OK;
+}
+
+status_t RTPReceiver::informSender(const sp<AMessage> ¶ms) {
+ if (!mRTCPConnected) {
+ return INVALID_OPERATION;
+ }
+
+ int64_t avgLatencyUs;
+ CHECK(params->findInt64("avgLatencyUs", &avgLatencyUs));
+
+ int64_t maxLatencyUs;
+ CHECK(params->findInt64("maxLatencyUs", &maxLatencyUs));
+
+ sp<ABuffer> buf = new ABuffer(28);
+
+ uint8_t *ptr = buf->data();
+ ptr[0] = 0x80 | 0;
+ ptr[1] = 204; // APP
+ ptr[2] = 0;
+
+ CHECK((buf->size() % 4) == 0u);
+ ptr[3] = (buf->size() / 4) - 1;
+
+ ptr[4] = kSourceID >> 24; // SSRC
+ ptr[5] = (kSourceID >> 16) & 0xff;
+ ptr[6] = (kSourceID >> 8) & 0xff;
+ ptr[7] = kSourceID & 0xff;
+ ptr[8] = 'l';
+ ptr[9] = 'a';
+ ptr[10] = 't';
+ ptr[11] = 'e';
+
+ ptr[12] = avgLatencyUs >> 56;
+ ptr[13] = (avgLatencyUs >> 48) & 0xff;
+ ptr[14] = (avgLatencyUs >> 40) & 0xff;
+ ptr[15] = (avgLatencyUs >> 32) & 0xff;
+ ptr[16] = (avgLatencyUs >> 24) & 0xff;
+ ptr[17] = (avgLatencyUs >> 16) & 0xff;
+ ptr[18] = (avgLatencyUs >> 8) & 0xff;
+ ptr[19] = avgLatencyUs & 0xff;
+
+ ptr[20] = maxLatencyUs >> 56;
+ ptr[21] = (maxLatencyUs >> 48) & 0xff;
+ ptr[22] = (maxLatencyUs >> 40) & 0xff;
+ ptr[23] = (maxLatencyUs >> 32) & 0xff;
+ ptr[24] = (maxLatencyUs >> 24) & 0xff;
+ ptr[25] = (maxLatencyUs >> 16) & 0xff;
+ ptr[26] = (maxLatencyUs >> 8) & 0xff;
+ ptr[27] = maxLatencyUs & 0xff;
+
+ mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size());
+
+ return OK;
+}
+
+void RTPReceiver::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatRTPNotify:
+ case kWhatRTCPNotify:
+ onNetNotify(msg->what() == kWhatRTPNotify, msg);
+ break;
+
+ case kWhatSendRR:
+ {
+ onSendRR();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void RTPReceiver::onNetNotify(bool isRTP, const sp<AMessage> &msg) {
+ int32_t reason;
+ CHECK(msg->findInt32("reason", &reason));
+
+ switch (reason) {
+ case ANetworkSession::kWhatError:
+ {
+ int32_t sessionID;
+ CHECK(msg->findInt32("sessionID", &sessionID));
+
+ int32_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ int32_t errorOccuredDuringSend;
+ CHECK(msg->findInt32("send", &errorOccuredDuringSend));
+
+ AString detail;
+ CHECK(msg->findString("detail", &detail));
+
+ ALOGE("An error occurred during %s in session %d "
+ "(%d, '%s' (%s)).",
+ errorOccuredDuringSend ? "send" : "receive",
+ sessionID,
+ err,
+ detail.c_str(),
+ strerror(-err));
+
+ mNetSession->destroySession(sessionID);
+
+ if (sessionID == mRTPSessionID) {
+ mRTPSessionID = 0;
+ } else if (sessionID == mRTCPSessionID) {
+ mRTCPSessionID = 0;
+ } else if (sessionID == mRTPClientSessionID) {
+ mRTPClientSessionID = 0;
+ } else if (sessionID == mRTCPClientSessionID) {
+ mRTCPClientSessionID = 0;
+ }
+
+ if (!mRTPConnected
+ || (mRTCPMode != TRANSPORT_NONE && !mRTCPConnected)) {
+ notifyInitDone(err);
+ break;
+ }
+
+ notifyError(err);
+ break;
+ }
+
+ case ANetworkSession::kWhatDatagram:
+ {
+ sp<ABuffer> data;
+ CHECK(msg->findBuffer("data", &data));
+
+ if (isRTP) {
+ if (mFlags & FLAG_AUTO_CONNECT) {
+ AString fromAddr;
+ CHECK(msg->findString("fromAddr", &fromAddr));
+
+ int32_t fromPort;
+ CHECK(msg->findInt32("fromPort", &fromPort));
+
+ CHECK_EQ((status_t)OK,
+ connect(
+ fromAddr.c_str(), fromPort, fromPort + 1));
+
+ mFlags &= ~FLAG_AUTO_CONNECT;
+ }
+
+ onRTPData(data);
+ } else {
+ onRTCPData(data);
+ }
+ break;
+ }
+
+ case ANetworkSession::kWhatClientConnected:
+ {
+ int32_t sessionID;
+ CHECK(msg->findInt32("sessionID", &sessionID));
+
+ if (isRTP) {
+ CHECK_EQ(mRTPMode, TRANSPORT_TCP);
+
+ if (mRTPClientSessionID != 0) {
+ // We only allow a single client connection.
+ mNetSession->destroySession(sessionID);
+ sessionID = 0;
+ break;
+ }
+
+ mRTPClientSessionID = sessionID;
+ mRTPConnected = true;
+ } else {
+ CHECK_EQ(mRTCPMode, TRANSPORT_TCP);
+
+ if (mRTCPClientSessionID != 0) {
+ // We only allow a single client connection.
+ mNetSession->destroySession(sessionID);
+ sessionID = 0;
+ break;
+ }
+
+ mRTCPClientSessionID = sessionID;
+ mRTCPConnected = true;
+ }
+
+ if (mRTPConnected
+ && (mRTCPConnected || mRTCPMode == TRANSPORT_NONE)) {
+ notifyInitDone(OK);
+ }
+ break;
+ }
+ }
+}
+
+void RTPReceiver::notifyInitDone(status_t err) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatInitDone);
+ notify->setInt32("err", err);
+ notify->post();
+}
+
+void RTPReceiver::notifyError(status_t err) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatError);
+ notify->setInt32("err", err);
+ notify->post();
+}
+
+void RTPReceiver::notifyPacketLost() {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatPacketLost);
+ notify->post();
+}
+
+status_t RTPReceiver::onRTPData(const sp<ABuffer> &buffer) {
+ size_t size = buffer->size();
+ if (size < 12) {
+ // Too short to be a valid RTP header.
+ return ERROR_MALFORMED;
+ }
+
+ const uint8_t *data = buffer->data();
+
+ if ((data[0] >> 6) != 2) {
+ // Unsupported version.
+ return ERROR_UNSUPPORTED;
+ }
+
+ if (data[0] & 0x20) {
+ // Padding present.
+
+ size_t paddingLength = data[size - 1];
+
+ if (paddingLength + 12 > size) {
+ // If we removed this much padding we'd end up with something
+ // that's too short to be a valid RTP header.
+ return ERROR_MALFORMED;
+ }
+
+ size -= paddingLength;
+ }
+
+ int numCSRCs = data[0] & 0x0f;
+
+ size_t payloadOffset = 12 + 4 * numCSRCs;
+
+ if (size < payloadOffset) {
+ // Not enough data to fit the basic header and all the CSRC entries.
+ return ERROR_MALFORMED;
+ }
+
+ if (data[0] & 0x10) {
+ // Header eXtension present.
+
+ if (size < payloadOffset + 4) {
+ // Not enough data to fit the basic header, all CSRC entries
+ // and the first 4 bytes of the extension header.
+
+ return ERROR_MALFORMED;
+ }
+
+ const uint8_t *extensionData = &data[payloadOffset];
+
+ size_t extensionLength =
+ 4 * (extensionData[2] << 8 | extensionData[3]);
+
+ if (size < payloadOffset + 4 + extensionLength) {
+ return ERROR_MALFORMED;
+ }
+
+ payloadOffset += 4 + extensionLength;
+ }
+
+ uint32_t srcId = U32_AT(&data[8]);
+ uint32_t rtpTime = U32_AT(&data[4]);
+ uint16_t seqNo = U16_AT(&data[2]);
+
+ sp<AMessage> meta = buffer->meta();
+ meta->setInt32("ssrc", srcId);
+ meta->setInt32("rtp-time", rtpTime);
+ meta->setInt32("PT", data[1] & 0x7f);
+ meta->setInt32("M", data[1] >> 7);
+
+ buffer->setRange(payloadOffset, size - payloadOffset);
+
+ ssize_t index = mSources.indexOfKey(srcId);
+ sp<Source> source;
+ if (index < 0) {
+ source = new Source(this, srcId);
+ looper()->registerHandler(source);
+
+ mSources.add(srcId, source);
+ } else {
+ source = mSources.valueAt(index);
+ }
+
+ source->onPacketReceived(seqNo, buffer);
+
+ return OK;
+}
+
+status_t RTPReceiver::onRTCPData(const sp<ABuffer> &data) {
+ ALOGI("onRTCPData");
+ return OK;
+}
+
+void RTPReceiver::addSDES(const sp<ABuffer> &buffer) {
+ uint8_t *data = buffer->data() + buffer->size();
+ data[0] = 0x80 | 1;
+ data[1] = 202; // SDES
+ data[4] = kSourceID >> 24; // SSRC
+ data[5] = (kSourceID >> 16) & 0xff;
+ data[6] = (kSourceID >> 8) & 0xff;
+ data[7] = kSourceID & 0xff;
+
+ size_t offset = 8;
+
+ data[offset++] = 1; // CNAME
+
+ AString cname = "stagefright@somewhere";
+ data[offset++] = cname.size();
+
+ memcpy(&data[offset], cname.c_str(), cname.size());
+ offset += cname.size();
+
+ data[offset++] = 6; // TOOL
+
+ AString tool = "stagefright/1.0";
+ data[offset++] = tool.size();
+
+ memcpy(&data[offset], tool.c_str(), tool.size());
+ offset += tool.size();
+
+ data[offset++] = 0;
+
+ if ((offset % 4) > 0) {
+ size_t count = 4 - (offset % 4);
+ switch (count) {
+ case 3:
+ data[offset++] = 0;
+ case 2:
+ data[offset++] = 0;
+ case 1:
+ data[offset++] = 0;
+ }
+ }
+
+ size_t numWords = (offset / 4) - 1;
+ data[2] = numWords >> 8;
+ data[3] = numWords & 0xff;
+
+ buffer->setRange(buffer->offset(), buffer->size() + offset);
+}
+
+void RTPReceiver::scheduleSendRR() {
+ (new AMessage(kWhatSendRR, id()))->post(5000000ll);
+}
+
+void RTPReceiver::onSendRR() {
+ sp<ABuffer> buf = new ABuffer(kMaxUDPPacketSize);
+ buf->setRange(0, 0);
+
+ uint8_t *ptr = buf->data();
+ ptr[0] = 0x80 | 0;
+ ptr[1] = 201; // RR
+ ptr[2] = 0;
+ ptr[3] = 1;
+ ptr[4] = kSourceID >> 24; // SSRC
+ ptr[5] = (kSourceID >> 16) & 0xff;
+ ptr[6] = (kSourceID >> 8) & 0xff;
+ ptr[7] = kSourceID & 0xff;
+
+ buf->setRange(0, 8);
+
+ size_t numReportBlocks = 0;
+ for (size_t i = 0; i < mSources.size(); ++i) {
+ uint32_t ssrc = mSources.keyAt(i);
+ sp<Source> source = mSources.valueAt(i);
+
+ if (numReportBlocks > 31 || buf->size() + 24 > buf->capacity()) {
+ // Cannot fit another report block.
+ break;
+ }
+
+ source->addReportBlock(ssrc, buf);
+ ++numReportBlocks;
+ }
+
+ ptr[0] |= numReportBlocks; // 5 bit
+
+ size_t sizeInWordsMinus1 = 1 + 6 * numReportBlocks;
+ ptr[2] = sizeInWordsMinus1 >> 8;
+ ptr[3] = sizeInWordsMinus1 & 0xff;
+
+ buf->setRange(0, (sizeInWordsMinus1 + 1) * 4);
+
+ addSDES(buf);
+
+ mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size());
+
+ scheduleSendRR();
+}
+
+status_t RTPReceiver::registerPacketType(
+ uint8_t packetType, PacketizationMode mode) {
+ mPacketTypes.add(packetType, mode);
+
+ return OK;
+}
+
+sp<RTPReceiver::Assembler> RTPReceiver::makeAssembler(uint8_t packetType) {
+ ssize_t index = mPacketTypes.indexOfKey(packetType);
+ if (index < 0) {
+ return NULL;
+ }
+
+ PacketizationMode mode = mPacketTypes.valueAt(index);
+
+ switch (mode) {
+ case PACKETIZATION_NONE:
+ case PACKETIZATION_TRANSPORT_STREAM:
+ return new TSAssembler(mNotify);
+
+ case PACKETIZATION_H264:
+ return new H264Assembler(mNotify);
+
+ default:
+ return NULL;
+ }
+}
+
+void RTPReceiver::requestRetransmission(uint32_t senderSSRC, int32_t extSeqNo) {
+ int32_t blp = 0;
+
+ sp<ABuffer> buf = new ABuffer(16);
+ buf->setRange(0, 0);
+
+ uint8_t *ptr = buf->data();
+ ptr[0] = 0x80 | 1; // generic NACK
+ ptr[1] = 205; // TSFB
+ ptr[2] = 0;
+ ptr[3] = 3;
+ ptr[8] = (senderSSRC >> 24) & 0xff;
+ ptr[9] = (senderSSRC >> 16) & 0xff;
+ ptr[10] = (senderSSRC >> 8) & 0xff;
+ ptr[11] = (senderSSRC & 0xff);
+ ptr[8] = (kSourceID >> 24) & 0xff;
+ ptr[9] = (kSourceID >> 16) & 0xff;
+ ptr[10] = (kSourceID >> 8) & 0xff;
+ ptr[11] = (kSourceID & 0xff);
+ ptr[12] = (extSeqNo >> 8) & 0xff;
+ ptr[13] = (extSeqNo & 0xff);
+ ptr[14] = (blp >> 8) & 0xff;
+ ptr[15] = (blp & 0xff);
+
+ buf->setRange(0, 16);
+
+ mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size());
+}
+
+void RTPReceiver::Source::modifyPacketStatus(int32_t extSeqNo, uint32_t mask) {
+#if TRACK_PACKET_LOSS
+ ssize_t index = mLostPackets.indexOfKey(extSeqNo);
+ if (index < 0) {
+ mLostPackets.add(extSeqNo, mask);
+ } else {
+ mLostPackets.editValueAt(index) |= mask;
+ }
+#endif
+}
+
+void RTPReceiver::Source::postRetransmitTimer(int64_t timeUs) {
+ int64_t delayUs = timeUs - ALooper::GetNowUs();
+ sp<AMessage> msg = new AMessage(kWhatRetransmit, id());
+ msg->setInt32("generation", mRetransmitGeneration);
+ msg->post(delayUs);
+}
+
+void RTPReceiver::Source::postDeclareLostTimer(int64_t timeUs) {
+ CHECK(!mDeclareLostTimerPending);
+ mDeclareLostTimerPending = true;
+
+ int64_t delayUs = timeUs - ALooper::GetNowUs();
+ sp<AMessage> msg = new AMessage(kWhatDeclareLost, id());
+ msg->setInt32("generation", mDeclareLostGeneration);
+ msg->post(delayUs);
+}
+
+void RTPReceiver::Source::cancelTimers() {
+ ++mRetransmitGeneration;
+ ++mDeclareLostGeneration;
+ mDeclareLostTimerPending = false;
+}
+
+} // namespace android
+
diff --git a/media/libstagefright/wifi-display/rtp/RTPReceiver.h b/media/libstagefright/wifi-display/rtp/RTPReceiver.h
new file mode 100644
index 0000000..240ab2e
--- /dev/null
+++ b/media/libstagefright/wifi-display/rtp/RTPReceiver.h
@@ -0,0 +1,125 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef RTP_RECEIVER_H_
+
+#define RTP_RECEIVER_H_
+
+#include "RTPBase.h"
+
+#include <media/stagefright/foundation/AHandler.h>
+
+namespace android {
+
+struct ABuffer;
+struct ANetworkSession;
+
+// An object of this class facilitates receiving of media data on an RTP
+// channel. The channel is established over a UDP or TCP connection depending
+// on which "TransportMode" was chosen. In addition different RTP packetization
+// schemes are supported such as "Transport Stream Packets over RTP",
+// or "AVC/H.264 encapsulation as specified in RFC 3984 (non-interleaved mode)"
+struct RTPReceiver : public RTPBase, public AHandler {
+ enum {
+ kWhatInitDone,
+ kWhatError,
+ kWhatAccessUnit,
+ kWhatPacketLost,
+ };
+
+ enum Flags {
+ FLAG_AUTO_CONNECT = 1,
+ };
+ RTPReceiver(
+ const sp<ANetworkSession> &netSession,
+ const sp<AMessage> ¬ify,
+ uint32_t flags = 0);
+
+ status_t registerPacketType(
+ uint8_t packetType, PacketizationMode mode);
+
+ status_t initAsync(
+ TransportMode rtpMode,
+ TransportMode rtcpMode,
+ int32_t *outLocalRTPPort);
+
+ status_t connect(
+ const char *remoteHost,
+ int32_t remoteRTPPort,
+ int32_t remoteRTCPPort);
+
+ status_t informSender(const sp<AMessage> ¶ms);
+
+protected:
+ virtual ~RTPReceiver();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kWhatRTPNotify,
+ kWhatRTCPNotify,
+ kWhatSendRR,
+ };
+
+ enum {
+ kSourceID = 0xdeadbeef,
+ kPacketLostAfterUs = 100000,
+ kRequestRetransmissionAfterUs = -1,
+ };
+
+ struct Assembler;
+ struct H264Assembler;
+ struct Source;
+ struct TSAssembler;
+
+ sp<ANetworkSession> mNetSession;
+ sp<AMessage> mNotify;
+ uint32_t mFlags;
+ TransportMode mRTPMode;
+ TransportMode mRTCPMode;
+ int32_t mRTPSessionID;
+ int32_t mRTCPSessionID;
+ bool mRTPConnected;
+ bool mRTCPConnected;
+
+ int32_t mRTPClientSessionID; // in TRANSPORT_TCP mode.
+ int32_t mRTCPClientSessionID; // in TRANSPORT_TCP mode.
+
+ KeyedVector<uint8_t, PacketizationMode> mPacketTypes;
+ KeyedVector<uint32_t, sp<Source> > mSources;
+
+ void onNetNotify(bool isRTP, const sp<AMessage> &msg);
+ status_t onRTPData(const sp<ABuffer> &data);
+ status_t onRTCPData(const sp<ABuffer> &data);
+ void onSendRR();
+
+ void scheduleSendRR();
+ void addSDES(const sp<ABuffer> &buffer);
+
+ void notifyInitDone(status_t err);
+ void notifyError(status_t err);
+ void notifyPacketLost();
+
+ sp<Assembler> makeAssembler(uint8_t packetType);
+
+ void requestRetransmission(uint32_t senderSSRC, int32_t extSeqNo);
+
+ DISALLOW_EVIL_CONSTRUCTORS(RTPReceiver);
+};
+
+} // namespace android
+
+#endif // RTP_RECEIVER_H_
diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.cpp b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
index 095fd97..6bbe650 100644
--- a/media/libstagefright/wifi-display/rtp/RTPSender.cpp
+++ b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
@@ -767,6 +767,17 @@
}
status_t RTPSender::parseAPP(const uint8_t *data, size_t size) {
+ if (!memcmp("late", &data[8], 4)) {
+ int64_t avgLatencyUs = (int64_t)U64_AT(&data[12]);
+ int64_t maxLatencyUs = (int64_t)U64_AT(&data[20]);
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatInformSender);
+ notify->setInt64("avgLatencyUs", avgLatencyUs);
+ notify->setInt64("maxLatencyUs", maxLatencyUs);
+ notify->post();
+ }
+
return OK;
}
diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.h b/media/libstagefright/wifi-display/rtp/RTPSender.h
index 7dc138a..fefcab7 100644
--- a/media/libstagefright/wifi-display/rtp/RTPSender.h
+++ b/media/libstagefright/wifi-display/rtp/RTPSender.h
@@ -37,6 +37,7 @@
kWhatInitDone,
kWhatError,
kWhatNetworkStall,
+ kWhatInformSender,
};
RTPSender(
const sp<ANetworkSession> &netSession,
diff --git a/media/libstagefright/wifi-display/rtptest.cpp b/media/libstagefright/wifi-display/rtptest.cpp
new file mode 100644
index 0000000..764a38b
--- /dev/null
+++ b/media/libstagefright/wifi-display/rtptest.cpp
@@ -0,0 +1,565 @@
+/*
+ * Copyright 2013, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NEBUG 0
+#define LOG_TAG "rtptest"
+#include <utils/Log.h>
+
+#include "ANetworkSession.h"
+#include "rtp/RTPSender.h"
+#include "rtp/RTPReceiver.h"
+#include "TimeSyncer.h"
+
+#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/NuMediaExtractor.h>
+#include <media/stagefright/Utils.h>
+
+#define MEDIA_FILENAME "/sdcard/Frame Counter HD 30FPS_1080p.mp4"
+
+namespace android {
+
+struct PacketSource : public RefBase {
+ PacketSource() {}
+
+ virtual sp<ABuffer> getNextAccessUnit() = 0;
+
+protected:
+ virtual ~PacketSource() {}
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(PacketSource);
+};
+
+struct MediaPacketSource : public PacketSource {
+ MediaPacketSource()
+ : mMaxSampleSize(1024 * 1024) {
+ mExtractor = new NuMediaExtractor;
+ CHECK_EQ((status_t)OK,
+ mExtractor->setDataSource(MEDIA_FILENAME));
+
+ bool haveVideo = false;
+ for (size_t i = 0; i < mExtractor->countTracks(); ++i) {
+ sp<AMessage> format;
+ CHECK_EQ((status_t)OK, mExtractor->getTrackFormat(i, &format));
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime.c_str())) {
+ mExtractor->selectTrack(i);
+ haveVideo = true;
+ break;
+ }
+ }
+
+ CHECK(haveVideo);
+ }
+
+ virtual sp<ABuffer> getNextAccessUnit() {
+ int64_t timeUs;
+ status_t err = mExtractor->getSampleTime(&timeUs);
+
+ if (err != OK) {
+ return NULL;
+ }
+
+ sp<ABuffer> accessUnit = new ABuffer(mMaxSampleSize);
+ CHECK_EQ((status_t)OK, mExtractor->readSampleData(accessUnit));
+
+ accessUnit->meta()->setInt64("timeUs", timeUs);
+
+ CHECK_EQ((status_t)OK, mExtractor->advance());
+
+ return accessUnit;
+ }
+
+protected:
+ virtual ~MediaPacketSource() {
+ }
+
+private:
+ sp<NuMediaExtractor> mExtractor;
+ size_t mMaxSampleSize;
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaPacketSource);
+};
+
+struct SimplePacketSource : public PacketSource {
+ SimplePacketSource()
+ : mCounter(0) {
+ }
+
+ virtual sp<ABuffer> getNextAccessUnit() {
+ sp<ABuffer> buffer = new ABuffer(4);
+ uint8_t *dst = buffer->data();
+ dst[0] = mCounter >> 24;
+ dst[1] = (mCounter >> 16) & 0xff;
+ dst[2] = (mCounter >> 8) & 0xff;
+ dst[3] = mCounter & 0xff;
+
+ buffer->meta()->setInt64("timeUs", mCounter * 1000000ll / kFrameRate);
+
+ ++mCounter;
+
+ return buffer;
+ }
+
+protected:
+ virtual ~SimplePacketSource() {
+ }
+
+private:
+ enum {
+ kFrameRate = 30
+ };
+
+ uint32_t mCounter;
+
+ DISALLOW_EVIL_CONSTRUCTORS(SimplePacketSource);
+};
+
+struct TestHandler : public AHandler {
+ TestHandler(const sp<ANetworkSession> &netSession);
+
+ void listen();
+ void connect(const char *host, int32_t port);
+
+protected:
+ virtual ~TestHandler();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kWhatListen,
+ kWhatConnect,
+ kWhatReceiverNotify,
+ kWhatSenderNotify,
+ kWhatSendMore,
+ kWhatStop,
+ kWhatTimeSyncerNotify,
+ };
+
+#if 1
+ static const RTPBase::TransportMode kRTPMode = RTPBase::TRANSPORT_UDP;
+ static const RTPBase::TransportMode kRTCPMode = RTPBase::TRANSPORT_UDP;
+#else
+ static const RTPBase::TransportMode kRTPMode = RTPBase::TRANSPORT_TCP;
+ static const RTPBase::TransportMode kRTCPMode = RTPBase::TRANSPORT_NONE;
+#endif
+
+#if 1
+ static const RTPBase::PacketizationMode kPacketizationMode
+ = RTPBase::PACKETIZATION_H264;
+#else
+ static const RTPBase::PacketizationMode kPacketizationMode
+ = RTPBase::PACKETIZATION_NONE;
+#endif
+
+ sp<ANetworkSession> mNetSession;
+ sp<PacketSource> mSource;
+ sp<RTPSender> mSender;
+ sp<RTPReceiver> mReceiver;
+
+ sp<TimeSyncer> mTimeSyncer;
+ bool mTimeSyncerStarted;
+
+ int64_t mFirstTimeRealUs;
+ int64_t mFirstTimeMediaUs;
+
+ int64_t mTimeOffsetUs;
+ bool mTimeOffsetValid;
+
+ status_t readMore();
+
+ DISALLOW_EVIL_CONSTRUCTORS(TestHandler);
+};
+
+TestHandler::TestHandler(const sp<ANetworkSession> &netSession)
+ : mNetSession(netSession),
+ mTimeSyncerStarted(false),
+ mFirstTimeRealUs(-1ll),
+ mFirstTimeMediaUs(-1ll),
+ mTimeOffsetUs(-1ll),
+ mTimeOffsetValid(false) {
+}
+
+TestHandler::~TestHandler() {
+}
+
+void TestHandler::listen() {
+ sp<AMessage> msg = new AMessage(kWhatListen, id());
+ msg->post();
+}
+
+void TestHandler::connect(const char *host, int32_t port) {
+ sp<AMessage> msg = new AMessage(kWhatConnect, id());
+ msg->setString("host", host);
+ msg->setInt32("port", port);
+ msg->post();
+}
+
+static void dumpDelay(int64_t delayMs) {
+ static const int64_t kMinDelayMs = 0;
+ static const int64_t kMaxDelayMs = 300;
+
+ const char *kPattern = "########################################";
+ size_t kPatternSize = strlen(kPattern);
+
+ int n = (kPatternSize * (delayMs - kMinDelayMs))
+ / (kMaxDelayMs - kMinDelayMs);
+
+ if (n < 0) {
+ n = 0;
+ } else if ((size_t)n > kPatternSize) {
+ n = kPatternSize;
+ }
+
+ ALOGI("(%4lld ms) %s\n",
+ delayMs,
+ kPattern + kPatternSize - n);
+}
+
+void TestHandler::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatListen:
+ {
+ sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
+ mTimeSyncer = new TimeSyncer(mNetSession, notify);
+ looper()->registerHandler(mTimeSyncer);
+
+ notify = new AMessage(kWhatReceiverNotify, id());
+ mReceiver = new RTPReceiver(
+ mNetSession, notify, RTPReceiver::FLAG_AUTO_CONNECT);
+ looper()->registerHandler(mReceiver);
+
+ CHECK_EQ((status_t)OK,
+ mReceiver->registerPacketType(33, kPacketizationMode));
+
+ int32_t receiverRTPPort;
+ CHECK_EQ((status_t)OK,
+ mReceiver->initAsync(
+ kRTPMode,
+ kRTCPMode,
+ &receiverRTPPort));
+
+ printf("picked receiverRTPPort %d\n", receiverRTPPort);
+
+#if 0
+ CHECK_EQ((status_t)OK,
+ mReceiver->connect(
+ "127.0.0.1", senderRTPPort, senderRTPPort + 1));
+#endif
+ break;
+ }
+
+ case kWhatConnect:
+ {
+ AString host;
+ CHECK(msg->findString("host", &host));
+
+ sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
+ mTimeSyncer = new TimeSyncer(mNetSession, notify);
+ looper()->registerHandler(mTimeSyncer);
+ mTimeSyncer->startServer(8123);
+
+ int32_t receiverRTPPort;
+ CHECK(msg->findInt32("port", &receiverRTPPort));
+
+#if 1
+ mSource = new MediaPacketSource;
+#else
+ mSource = new SimplePacketSource;
+#endif
+
+ notify = new AMessage(kWhatSenderNotify, id());
+ mSender = new RTPSender(mNetSession, notify);
+
+ looper()->registerHandler(mSender);
+
+ int32_t senderRTPPort;
+ CHECK_EQ((status_t)OK,
+ mSender->initAsync(
+ host.c_str(),
+ receiverRTPPort,
+ kRTPMode,
+ kRTCPMode == RTPBase::TRANSPORT_NONE
+ ? -1 : receiverRTPPort + 1,
+ kRTCPMode,
+ &senderRTPPort));
+
+ printf("picked senderRTPPort %d\n", senderRTPPort);
+ break;
+ }
+
+ case kWhatSenderNotify:
+ {
+ ALOGI("kWhatSenderNotify");
+
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case RTPSender::kWhatInitDone:
+ {
+ int32_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ ALOGI("RTPSender::initAsync completed w/ err %d", err);
+
+ if (err == OK) {
+ err = readMore();
+
+ if (err != OK) {
+ (new AMessage(kWhatStop, id()))->post();
+ }
+ }
+ break;
+ }
+
+ case RTPSender::kWhatError:
+ break;
+ }
+ break;
+ }
+
+ case kWhatReceiverNotify:
+ {
+ ALOGV("kWhatReceiverNotify");
+
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case RTPReceiver::kWhatInitDone:
+ {
+ int32_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ ALOGI("RTPReceiver::initAsync completed w/ err %d", err);
+ break;
+ }
+
+ case RTPReceiver::kWhatError:
+ break;
+
+ case RTPReceiver::kWhatAccessUnit:
+ {
+#if 0
+ if (!mTimeSyncerStarted) {
+ mTimeSyncer->startClient("172.18.41.216", 8123);
+ mTimeSyncerStarted = true;
+ }
+
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("accessUnit", &accessUnit));
+
+ int64_t timeUs;
+ CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+
+ if (mTimeOffsetValid) {
+ timeUs -= mTimeOffsetUs;
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t delayMs = (nowUs - timeUs) / 1000ll;
+
+ dumpDelay(delayMs);
+ }
+#endif
+ break;
+ }
+
+ case RTPReceiver::kWhatPacketLost:
+ ALOGV("kWhatPacketLost");
+ break;
+
+ default:
+ TRESPASS();
+ }
+ break;
+ }
+
+ case kWhatSendMore:
+ {
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("accessUnit", &accessUnit));
+
+ CHECK_EQ((status_t)OK,
+ mSender->queueBuffer(
+ accessUnit,
+ 33,
+ kPacketizationMode));
+
+ status_t err = readMore();
+
+ if (err != OK) {
+ (new AMessage(kWhatStop, id()))->post();
+ }
+ break;
+ }
+
+ case kWhatStop:
+ {
+ if (mReceiver != NULL) {
+ looper()->unregisterHandler(mReceiver->id());
+ mReceiver.clear();
+ }
+
+ if (mSender != NULL) {
+ looper()->unregisterHandler(mSender->id());
+ mSender.clear();
+ }
+
+ mSource.clear();
+
+ looper()->stop();
+ break;
+ }
+
+ case kWhatTimeSyncerNotify:
+ {
+ CHECK(msg->findInt64("offset", &mTimeOffsetUs));
+ mTimeOffsetValid = true;
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+status_t TestHandler::readMore() {
+ sp<ABuffer> accessUnit = mSource->getNextAccessUnit();
+
+ if (accessUnit == NULL) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ int64_t timeUs;
+ CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t whenUs;
+
+ if (mFirstTimeRealUs < 0ll) {
+ mFirstTimeRealUs = whenUs = nowUs;
+ mFirstTimeMediaUs = timeUs;
+ } else {
+ whenUs = mFirstTimeRealUs + timeUs - mFirstTimeMediaUs;
+ }
+
+ accessUnit->meta()->setInt64("timeUs", whenUs);
+
+ sp<AMessage> msg = new AMessage(kWhatSendMore, id());
+ msg->setBuffer("accessUnit", accessUnit);
+ msg->post(whenUs - nowUs);
+
+ return OK;
+}
+
+} // namespace android
+
+static void usage(const char *me) {
+ fprintf(stderr,
+ "usage: %s -c host:port\tconnect to remote host\n"
+ " -l \tlisten\n",
+ me);
+}
+
+int main(int argc, char **argv) {
+ using namespace android;
+
+ // srand(time(NULL));
+
+ ProcessState::self()->startThreadPool();
+
+ DataSource::RegisterDefaultSniffers();
+
+ bool listen = false;
+ int32_t connectToPort = -1;
+ AString connectToHost;
+
+ int res;
+ while ((res = getopt(argc, argv, "hc:l")) >= 0) {
+ switch (res) {
+ case 'c':
+ {
+ const char *colonPos = strrchr(optarg, ':');
+
+ if (colonPos == NULL) {
+ usage(argv[0]);
+ exit(1);
+ }
+
+ connectToHost.setTo(optarg, colonPos - optarg);
+
+ char *end;
+ connectToPort = strtol(colonPos + 1, &end, 10);
+
+ if (*end != '\0' || end == colonPos + 1
+ || connectToPort < 1 || connectToPort > 65535) {
+ fprintf(stderr, "Illegal port specified.\n");
+ exit(1);
+ }
+ break;
+ }
+
+ case 'l':
+ {
+ listen = true;
+ break;
+ }
+
+ case '?':
+ case 'h':
+ usage(argv[0]);
+ exit(1);
+ }
+ }
+
+ if (!listen && connectToPort < 0) {
+ fprintf(stderr,
+ "You need to select either client or server mode.\n");
+ exit(1);
+ }
+
+ sp<ANetworkSession> netSession = new ANetworkSession;
+ netSession->start();
+
+ sp<ALooper> looper = new ALooper;
+
+ sp<TestHandler> handler = new TestHandler(netSession);
+ looper->registerHandler(handler);
+
+ if (listen) {
+ handler->listen();
+ }
+
+ if (connectToPort >= 0) {
+ handler->connect(connectToHost.c_str(), connectToPort);
+ }
+
+ looper->start(true /* runOnCallingThread */);
+
+ return 0;
+}
+
diff --git a/media/libstagefright/wifi-display/sink/DirectRenderer.cpp b/media/libstagefright/wifi-display/sink/DirectRenderer.cpp
new file mode 100644
index 0000000..15f9c88
--- /dev/null
+++ b/media/libstagefright/wifi-display/sink/DirectRenderer.cpp
@@ -0,0 +1,625 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "DirectRenderer"
+#include <utils/Log.h>
+
+#include "DirectRenderer.h"
+
+#include <gui/SurfaceComposerClient.h>
+#include <gui/Surface.h>
+#include <media/AudioTrack.h>
+#include <media/ICrypto.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+/*
+ Drives the decoding process using a MediaCodec instance. Input buffers
+ queued by calls to "queueInputBuffer" are fed to the decoder as soon
+ as the decoder is ready for them, the client is notified about output
+ buffers as the decoder spits them out.
+*/
+struct DirectRenderer::DecoderContext : public AHandler {
+ enum {
+ kWhatOutputBufferReady,
+ };
+ DecoderContext(const sp<AMessage> ¬ify);
+
+ status_t init(
+ const sp<AMessage> &format,
+ const sp<IGraphicBufferProducer> &surfaceTex);
+
+ void queueInputBuffer(const sp<ABuffer> &accessUnit);
+
+ status_t renderOutputBufferAndRelease(size_t index);
+ status_t releaseOutputBuffer(size_t index);
+
+protected:
+ virtual ~DecoderContext();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kWhatDecoderNotify,
+ };
+
+ sp<AMessage> mNotify;
+ sp<ALooper> mDecoderLooper;
+ sp<MediaCodec> mDecoder;
+ Vector<sp<ABuffer> > mDecoderInputBuffers;
+ Vector<sp<ABuffer> > mDecoderOutputBuffers;
+ List<size_t> mDecoderInputBuffersAvailable;
+ bool mDecoderNotificationPending;
+
+ List<sp<ABuffer> > mAccessUnits;
+
+ void onDecoderNotify();
+ void scheduleDecoderNotification();
+ void queueDecoderInputBuffers();
+
+ void queueOutputBuffer(
+ size_t index, int64_t timeUs, const sp<ABuffer> &buffer);
+
+ DISALLOW_EVIL_CONSTRUCTORS(DecoderContext);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+/*
+ A "push" audio renderer. The primary function of this renderer is to use
+ an AudioTrack in push mode and making sure not to block the event loop
+ be ensuring that calls to AudioTrack::write never block. This is done by
+ estimating an upper bound of data that can be written to the AudioTrack
+ buffer without delay.
+*/
+struct DirectRenderer::AudioRenderer : public AHandler {
+ AudioRenderer(const sp<DecoderContext> &decoderContext);
+
+ void queueInputBuffer(
+ size_t index, int64_t timeUs, const sp<ABuffer> &buffer);
+
+protected:
+ virtual ~AudioRenderer();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kWhatPushAudio,
+ };
+
+ struct BufferInfo {
+ size_t mIndex;
+ int64_t mTimeUs;
+ sp<ABuffer> mBuffer;
+ };
+
+ sp<DecoderContext> mDecoderContext;
+ sp<AudioTrack> mAudioTrack;
+
+ List<BufferInfo> mInputBuffers;
+ bool mPushPending;
+
+ size_t mNumFramesWritten;
+
+ void schedulePushIfNecessary();
+ void onPushAudio();
+
+ ssize_t writeNonBlocking(const uint8_t *data, size_t size);
+
+ DISALLOW_EVIL_CONSTRUCTORS(AudioRenderer);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+DirectRenderer::DecoderContext::DecoderContext(const sp<AMessage> ¬ify)
+ : mNotify(notify),
+ mDecoderNotificationPending(false) {
+}
+
+DirectRenderer::DecoderContext::~DecoderContext() {
+ if (mDecoder != NULL) {
+ mDecoder->release();
+ mDecoder.clear();
+
+ mDecoderLooper->stop();
+ mDecoderLooper.clear();
+ }
+}
+
+status_t DirectRenderer::DecoderContext::init(
+ const sp<AMessage> &format,
+ const sp<IGraphicBufferProducer> &surfaceTex) {
+ CHECK(mDecoder == NULL);
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ mDecoderLooper = new ALooper;
+ mDecoderLooper->setName("video codec looper");
+
+ mDecoderLooper->start(
+ false /* runOnCallingThread */,
+ false /* canCallJava */,
+ PRIORITY_DEFAULT);
+
+ mDecoder = MediaCodec::CreateByType(
+ mDecoderLooper, mime.c_str(), false /* encoder */);
+
+ CHECK(mDecoder != NULL);
+
+ status_t err = mDecoder->configure(
+ format,
+ surfaceTex == NULL
+ ? NULL : new Surface(surfaceTex),
+ NULL /* crypto */,
+ 0 /* flags */);
+ CHECK_EQ(err, (status_t)OK);
+
+ err = mDecoder->start();
+ CHECK_EQ(err, (status_t)OK);
+
+ err = mDecoder->getInputBuffers(
+ &mDecoderInputBuffers);
+ CHECK_EQ(err, (status_t)OK);
+
+ err = mDecoder->getOutputBuffers(
+ &mDecoderOutputBuffers);
+ CHECK_EQ(err, (status_t)OK);
+
+ scheduleDecoderNotification();
+
+ return OK;
+}
+
+void DirectRenderer::DecoderContext::queueInputBuffer(
+ const sp<ABuffer> &accessUnit) {
+ CHECK(mDecoder != NULL);
+
+ mAccessUnits.push_back(accessUnit);
+ queueDecoderInputBuffers();
+}
+
+status_t DirectRenderer::DecoderContext::renderOutputBufferAndRelease(
+ size_t index) {
+ return mDecoder->renderOutputBufferAndRelease(index);
+}
+
+status_t DirectRenderer::DecoderContext::releaseOutputBuffer(size_t index) {
+ return mDecoder->releaseOutputBuffer(index);
+}
+
+void DirectRenderer::DecoderContext::queueDecoderInputBuffers() {
+ if (mDecoder == NULL) {
+ return;
+ }
+
+ bool submittedMore = false;
+
+ while (!mAccessUnits.empty()
+ && !mDecoderInputBuffersAvailable.empty()) {
+ size_t index = *mDecoderInputBuffersAvailable.begin();
+
+ mDecoderInputBuffersAvailable.erase(
+ mDecoderInputBuffersAvailable.begin());
+
+ sp<ABuffer> srcBuffer = *mAccessUnits.begin();
+ mAccessUnits.erase(mAccessUnits.begin());
+
+ const sp<ABuffer> &dstBuffer =
+ mDecoderInputBuffers.itemAt(index);
+
+ memcpy(dstBuffer->data(), srcBuffer->data(), srcBuffer->size());
+
+ int64_t timeUs;
+ CHECK(srcBuffer->meta()->findInt64("timeUs", &timeUs));
+
+ status_t err = mDecoder->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ srcBuffer->size(),
+ timeUs,
+ 0 /* flags */);
+ CHECK_EQ(err, (status_t)OK);
+
+ submittedMore = true;
+ }
+
+ if (submittedMore) {
+ scheduleDecoderNotification();
+ }
+}
+
+void DirectRenderer::DecoderContext::onMessageReceived(
+ const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatDecoderNotify:
+ {
+ onDecoderNotify();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void DirectRenderer::DecoderContext::onDecoderNotify() {
+ mDecoderNotificationPending = false;
+
+ for (;;) {
+ size_t index;
+ status_t err = mDecoder->dequeueInputBuffer(&index);
+
+ if (err == OK) {
+ mDecoderInputBuffersAvailable.push_back(index);
+ } else if (err == -EAGAIN) {
+ break;
+ } else {
+ TRESPASS();
+ }
+ }
+
+ queueDecoderInputBuffers();
+
+ for (;;) {
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t timeUs;
+ uint32_t flags;
+ status_t err = mDecoder->dequeueOutputBuffer(
+ &index,
+ &offset,
+ &size,
+ &timeUs,
+ &flags);
+
+ if (err == OK) {
+ queueOutputBuffer(
+ index, timeUs, mDecoderOutputBuffers.itemAt(index));
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ err = mDecoder->getOutputBuffers(
+ &mDecoderOutputBuffers);
+ CHECK_EQ(err, (status_t)OK);
+ } else if (err == INFO_FORMAT_CHANGED) {
+ // We don't care.
+ } else if (err == -EAGAIN) {
+ break;
+ } else {
+ TRESPASS();
+ }
+ }
+
+ scheduleDecoderNotification();
+}
+
+void DirectRenderer::DecoderContext::scheduleDecoderNotification() {
+ if (mDecoderNotificationPending) {
+ return;
+ }
+
+ sp<AMessage> notify =
+ new AMessage(kWhatDecoderNotify, id());
+
+ mDecoder->requestActivityNotification(notify);
+ mDecoderNotificationPending = true;
+}
+
+void DirectRenderer::DecoderContext::queueOutputBuffer(
+ size_t index, int64_t timeUs, const sp<ABuffer> &buffer) {
+ sp<AMessage> msg = mNotify->dup();
+ msg->setInt32("what", kWhatOutputBufferReady);
+ msg->setSize("index", index);
+ msg->setInt64("timeUs", timeUs);
+ msg->setBuffer("buffer", buffer);
+ msg->post();
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+DirectRenderer::AudioRenderer::AudioRenderer(
+ const sp<DecoderContext> &decoderContext)
+ : mDecoderContext(decoderContext),
+ mPushPending(false),
+ mNumFramesWritten(0) {
+ mAudioTrack = new AudioTrack(
+ AUDIO_STREAM_DEFAULT,
+ 48000.0f,
+ AUDIO_FORMAT_PCM,
+ AUDIO_CHANNEL_OUT_STEREO,
+ (int)0 /* frameCount */);
+
+ CHECK_EQ((status_t)OK, mAudioTrack->initCheck());
+
+ mAudioTrack->start();
+}
+
+DirectRenderer::AudioRenderer::~AudioRenderer() {
+}
+
+void DirectRenderer::AudioRenderer::queueInputBuffer(
+ size_t index, int64_t timeUs, const sp<ABuffer> &buffer) {
+ BufferInfo info;
+ info.mIndex = index;
+ info.mTimeUs = timeUs;
+ info.mBuffer = buffer;
+
+ mInputBuffers.push_back(info);
+ schedulePushIfNecessary();
+}
+
+void DirectRenderer::AudioRenderer::onMessageReceived(
+ const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatPushAudio:
+ {
+ onPushAudio();
+ break;
+ }
+
+ default:
+ break;
+ }
+}
+
+void DirectRenderer::AudioRenderer::schedulePushIfNecessary() {
+ if (mPushPending || mInputBuffers.empty()) {
+ return;
+ }
+
+ mPushPending = true;
+
+ uint32_t numFramesPlayed;
+ CHECK_EQ(mAudioTrack->getPosition(&numFramesPlayed),
+ (status_t)OK);
+
+ uint32_t numFramesPendingPlayout = mNumFramesWritten - numFramesPlayed;
+
+ // This is how long the audio sink will have data to
+ // play back.
+ const float msecsPerFrame = 1000.0f / mAudioTrack->getSampleRate();
+
+ int64_t delayUs =
+ msecsPerFrame * numFramesPendingPlayout * 1000ll;
+
+ // Let's give it more data after about half that time
+ // has elapsed.
+ (new AMessage(kWhatPushAudio, id()))->post(delayUs / 2);
+}
+
+void DirectRenderer::AudioRenderer::onPushAudio() {
+ mPushPending = false;
+
+ while (!mInputBuffers.empty()) {
+ const BufferInfo &info = *mInputBuffers.begin();
+
+ ssize_t n = writeNonBlocking(
+ info.mBuffer->data(), info.mBuffer->size());
+
+ if (n < (ssize_t)info.mBuffer->size()) {
+ CHECK_GE(n, 0);
+
+ info.mBuffer->setRange(
+ info.mBuffer->offset() + n, info.mBuffer->size() - n);
+ break;
+ }
+
+ mDecoderContext->releaseOutputBuffer(info.mIndex);
+
+ mInputBuffers.erase(mInputBuffers.begin());
+ }
+
+ schedulePushIfNecessary();
+}
+
+ssize_t DirectRenderer::AudioRenderer::writeNonBlocking(
+ const uint8_t *data, size_t size) {
+ uint32_t numFramesPlayed;
+ status_t err = mAudioTrack->getPosition(&numFramesPlayed);
+ if (err != OK) {
+ return err;
+ }
+
+ ssize_t numFramesAvailableToWrite =
+ mAudioTrack->frameCount() - (mNumFramesWritten - numFramesPlayed);
+
+ size_t numBytesAvailableToWrite =
+ numFramesAvailableToWrite * mAudioTrack->frameSize();
+
+ if (size > numBytesAvailableToWrite) {
+ size = numBytesAvailableToWrite;
+ }
+
+ CHECK_EQ(mAudioTrack->write(data, size), (ssize_t)size);
+
+ size_t numFramesWritten = size / mAudioTrack->frameSize();
+ mNumFramesWritten += numFramesWritten;
+
+ return size;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+DirectRenderer::DirectRenderer(
+ const sp<IGraphicBufferProducer> &bufferProducer)
+ : mSurfaceTex(bufferProducer),
+ mVideoRenderPending(false),
+ mNumFramesLate(0),
+ mNumFrames(0) {
+}
+
+DirectRenderer::~DirectRenderer() {
+}
+
+void DirectRenderer::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatDecoderNotify:
+ {
+ onDecoderNotify(msg);
+ break;
+ }
+
+ case kWhatRenderVideo:
+ {
+ onRenderVideo();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void DirectRenderer::setFormat(size_t trackIndex, const sp<AMessage> &format) {
+ CHECK_LT(trackIndex, 2u);
+
+ CHECK(mDecoderContext[trackIndex] == NULL);
+
+ sp<AMessage> notify = new AMessage(kWhatDecoderNotify, id());
+ notify->setSize("trackIndex", trackIndex);
+
+ mDecoderContext[trackIndex] = new DecoderContext(notify);
+ looper()->registerHandler(mDecoderContext[trackIndex]);
+
+ CHECK_EQ((status_t)OK,
+ mDecoderContext[trackIndex]->init(
+ format, trackIndex == 0 ? mSurfaceTex : NULL));
+
+ if (trackIndex == 1) {
+ // Audio
+ mAudioRenderer = new AudioRenderer(mDecoderContext[1]);
+ looper()->registerHandler(mAudioRenderer);
+ }
+}
+
+void DirectRenderer::queueAccessUnit(
+ size_t trackIndex, const sp<ABuffer> &accessUnit) {
+ CHECK_LT(trackIndex, 2u);
+
+ if (mDecoderContext[trackIndex] == NULL) {
+ CHECK_EQ(trackIndex, 0u);
+
+ sp<AMessage> format = new AMessage;
+ format->setString("mime", "video/avc");
+ format->setInt32("width", 640);
+ format->setInt32("height", 360);
+
+ setFormat(trackIndex, format);
+ }
+
+ mDecoderContext[trackIndex]->queueInputBuffer(accessUnit);
+}
+
+void DirectRenderer::onDecoderNotify(const sp<AMessage> &msg) {
+ size_t trackIndex;
+ CHECK(msg->findSize("trackIndex", &trackIndex));
+
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case DecoderContext::kWhatOutputBufferReady:
+ {
+ size_t index;
+ CHECK(msg->findSize("index", &index));
+
+ int64_t timeUs;
+ CHECK(msg->findInt64("timeUs", &timeUs));
+
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
+
+ queueOutputBuffer(trackIndex, index, timeUs, buffer);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void DirectRenderer::queueOutputBuffer(
+ size_t trackIndex,
+ size_t index, int64_t timeUs, const sp<ABuffer> &buffer) {
+ if (trackIndex == 1) {
+ // Audio
+ mAudioRenderer->queueInputBuffer(index, timeUs, buffer);
+ return;
+ }
+
+ OutputInfo info;
+ info.mIndex = index;
+ info.mTimeUs = timeUs;
+ info.mBuffer = buffer;
+ mVideoOutputBuffers.push_back(info);
+
+ scheduleVideoRenderIfNecessary();
+}
+
+void DirectRenderer::scheduleVideoRenderIfNecessary() {
+ if (mVideoRenderPending || mVideoOutputBuffers.empty()) {
+ return;
+ }
+
+ mVideoRenderPending = true;
+
+ int64_t timeUs = (*mVideoOutputBuffers.begin()).mTimeUs;
+ int64_t nowUs = ALooper::GetNowUs();
+
+ int64_t delayUs = timeUs - nowUs;
+
+ (new AMessage(kWhatRenderVideo, id()))->post(delayUs);
+}
+
+void DirectRenderer::onRenderVideo() {
+ mVideoRenderPending = false;
+
+ int64_t nowUs = ALooper::GetNowUs();
+
+ while (!mVideoOutputBuffers.empty()) {
+ const OutputInfo &info = *mVideoOutputBuffers.begin();
+
+ if (info.mTimeUs > nowUs) {
+ break;
+ }
+
+ if (info.mTimeUs + 15000ll < nowUs) {
+ ++mNumFramesLate;
+ }
+ ++mNumFrames;
+
+ status_t err =
+ mDecoderContext[0]->renderOutputBufferAndRelease(info.mIndex);
+ CHECK_EQ(err, (status_t)OK);
+
+ mVideoOutputBuffers.erase(mVideoOutputBuffers.begin());
+ }
+
+ scheduleVideoRenderIfNecessary();
+}
+
+} // namespace android
+
diff --git a/media/libstagefright/wifi-display/sink/DirectRenderer.h b/media/libstagefright/wifi-display/sink/DirectRenderer.h
new file mode 100644
index 0000000..c5a4a83
--- /dev/null
+++ b/media/libstagefright/wifi-display/sink/DirectRenderer.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef DIRECT_RENDERER_H_
+
+#define DIRECT_RENDERER_H_
+
+#include <media/stagefright/foundation/AHandler.h>
+
+namespace android {
+
+struct ABuffer;
+struct AudioTrack;
+struct IGraphicBufferProducer;
+struct MediaCodec;
+
+// Renders audio and video data queued by calls to "queueAccessUnit".
+struct DirectRenderer : public AHandler {
+ DirectRenderer(const sp<IGraphicBufferProducer> &bufferProducer);
+
+ void setFormat(size_t trackIndex, const sp<AMessage> &format);
+ void queueAccessUnit(size_t trackIndex, const sp<ABuffer> &accessUnit);
+
+protected:
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+ virtual ~DirectRenderer();
+
+private:
+ struct DecoderContext;
+ struct AudioRenderer;
+
+ enum {
+ kWhatDecoderNotify,
+ kWhatRenderVideo,
+ };
+
+ struct OutputInfo {
+ size_t mIndex;
+ int64_t mTimeUs;
+ sp<ABuffer> mBuffer;
+ };
+
+ sp<IGraphicBufferProducer> mSurfaceTex;
+
+ sp<DecoderContext> mDecoderContext[2];
+ List<OutputInfo> mVideoOutputBuffers;
+
+ bool mVideoRenderPending;
+
+ sp<AudioRenderer> mAudioRenderer;
+
+ int32_t mNumFramesLate;
+ int32_t mNumFrames;
+
+ void onDecoderNotify(const sp<AMessage> &msg);
+
+ void queueOutputBuffer(
+ size_t trackIndex,
+ size_t index, int64_t timeUs, const sp<ABuffer> &buffer);
+
+ void scheduleVideoRenderIfNecessary();
+ void onRenderVideo();
+
+ DISALLOW_EVIL_CONSTRUCTORS(DirectRenderer);
+};
+
+} // namespace android
+
+#endif // DIRECT_RENDERER_H_
diff --git a/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp b/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp
new file mode 100644
index 0000000..5db2099
--- /dev/null
+++ b/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp
@@ -0,0 +1,917 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "WifiDisplaySink"
+#include <utils/Log.h>
+
+#include "WifiDisplaySink.h"
+
+#include "DirectRenderer.h"
+#include "MediaReceiver.h"
+#include "ParsedMessage.h"
+#include "TimeSyncer.h"
+
+#include <cutils/properties.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+// static
+const AString WifiDisplaySink::sUserAgent = MakeUserAgent();
+
+WifiDisplaySink::WifiDisplaySink(
+ uint32_t flags,
+ const sp<ANetworkSession> &netSession,
+ const sp<IGraphicBufferProducer> &bufferProducer,
+ const sp<AMessage> ¬ify)
+ : mState(UNDEFINED),
+ mFlags(flags),
+ mNetSession(netSession),
+ mSurfaceTex(bufferProducer),
+ mNotify(notify),
+ mUsingTCPTransport(false),
+ mUsingTCPInterleaving(false),
+ mSessionID(0),
+ mNextCSeq(1),
+ mIDRFrameRequestPending(false),
+ mTimeOffsetUs(0ll),
+ mTimeOffsetValid(false),
+ mSetupDeferred(false),
+ mLatencyCount(0),
+ mLatencySumUs(0ll),
+ mLatencyMaxUs(0ll),
+ mMaxDelayMs(-1ll) {
+ // We support any and all resolutions, but prefer 720p30
+ mSinkSupportedVideoFormats.setNativeResolution(
+ VideoFormats::RESOLUTION_CEA, 5); // 1280 x 720 p30
+
+ mSinkSupportedVideoFormats.enableAll();
+}
+
+WifiDisplaySink::~WifiDisplaySink() {
+}
+
+void WifiDisplaySink::start(const char *sourceHost, int32_t sourcePort) {
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+ msg->setString("sourceHost", sourceHost);
+ msg->setInt32("sourcePort", sourcePort);
+ msg->post();
+}
+
+void WifiDisplaySink::start(const char *uri) {
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+ msg->setString("setupURI", uri);
+ msg->post();
+}
+
+// static
+bool WifiDisplaySink::ParseURL(
+ const char *url, AString *host, int32_t *port, AString *path,
+ AString *user, AString *pass) {
+ host->clear();
+ *port = 0;
+ path->clear();
+ user->clear();
+ pass->clear();
+
+ if (strncasecmp("rtsp://", url, 7)) {
+ return false;
+ }
+
+ const char *slashPos = strchr(&url[7], '/');
+
+ if (slashPos == NULL) {
+ host->setTo(&url[7]);
+ path->setTo("/");
+ } else {
+ host->setTo(&url[7], slashPos - &url[7]);
+ path->setTo(slashPos);
+ }
+
+ ssize_t atPos = host->find("@");
+
+ if (atPos >= 0) {
+ // Split of user:pass@ from hostname.
+
+ AString userPass(*host, 0, atPos);
+ host->erase(0, atPos + 1);
+
+ ssize_t colonPos = userPass.find(":");
+
+ if (colonPos < 0) {
+ *user = userPass;
+ } else {
+ user->setTo(userPass, 0, colonPos);
+ pass->setTo(userPass, colonPos + 1, userPass.size() - colonPos - 1);
+ }
+ }
+
+ const char *colonPos = strchr(host->c_str(), ':');
+
+ if (colonPos != NULL) {
+ char *end;
+ unsigned long x = strtoul(colonPos + 1, &end, 10);
+
+ if (end == colonPos + 1 || *end != '\0' || x >= 65536) {
+ return false;
+ }
+
+ *port = x;
+
+ size_t colonOffset = colonPos - host->c_str();
+ size_t trailing = host->size() - colonOffset;
+ host->erase(colonOffset, trailing);
+ } else {
+ *port = 554;
+ }
+
+ return true;
+}
+
+void WifiDisplaySink::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatStart:
+ {
+ sleep(2); // XXX
+
+ int32_t sourcePort;
+ CHECK(msg->findString("sourceHost", &mRTSPHost));
+ CHECK(msg->findInt32("sourcePort", &sourcePort));
+
+ sp<AMessage> notify = new AMessage(kWhatRTSPNotify, id());
+
+ status_t err = mNetSession->createRTSPClient(
+ mRTSPHost.c_str(), sourcePort, notify, &mSessionID);
+ CHECK_EQ(err, (status_t)OK);
+
+ mState = CONNECTING;
+ break;
+ }
+
+ case kWhatRTSPNotify:
+ {
+ int32_t reason;
+ CHECK(msg->findInt32("reason", &reason));
+
+ switch (reason) {
+ case ANetworkSession::kWhatError:
+ {
+ int32_t sessionID;
+ CHECK(msg->findInt32("sessionID", &sessionID));
+
+ int32_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ AString detail;
+ CHECK(msg->findString("detail", &detail));
+
+ ALOGE("An error occurred in session %d (%d, '%s/%s').",
+ sessionID,
+ err,
+ detail.c_str(),
+ strerror(-err));
+
+ if (sessionID == mSessionID) {
+ ALOGI("Lost control connection.");
+
+ // The control connection is dead now.
+ mNetSession->destroySession(mSessionID);
+ mSessionID = 0;
+
+ if (mNotify == NULL) {
+ looper()->stop();
+ } else {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatDisconnected);
+ notify->post();
+ }
+ }
+ break;
+ }
+
+ case ANetworkSession::kWhatConnected:
+ {
+ ALOGI("We're now connected.");
+ mState = CONNECTED;
+
+ if (mFlags & FLAG_SPECIAL_MODE) {
+ sp<AMessage> notify = new AMessage(
+ kWhatTimeSyncerNotify, id());
+
+ mTimeSyncer = new TimeSyncer(mNetSession, notify);
+ looper()->registerHandler(mTimeSyncer);
+
+ mTimeSyncer->startClient(mRTSPHost.c_str(), 8123);
+ }
+ break;
+ }
+
+ case ANetworkSession::kWhatData:
+ {
+ onReceiveClientData(msg);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+ break;
+ }
+
+ case kWhatStop:
+ {
+ looper()->stop();
+ break;
+ }
+
+ case kWhatMediaReceiverNotify:
+ {
+ onMediaReceiverNotify(msg);
+ break;
+ }
+
+ case kWhatTimeSyncerNotify:
+ {
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ if (what == TimeSyncer::kWhatTimeOffset) {
+ CHECK(msg->findInt64("offset", &mTimeOffsetUs));
+ mTimeOffsetValid = true;
+
+ if (mSetupDeferred) {
+ CHECK_EQ((status_t)OK,
+ sendSetup(
+ mSessionID,
+ "rtsp://x.x.x.x:x/wfd1.0/streamid=0"));
+
+ mSetupDeferred = false;
+ }
+ }
+ break;
+ }
+
+ case kWhatReportLateness:
+ {
+ if (mLatencyCount > 0) {
+ int64_t avgLatencyUs = mLatencySumUs / mLatencyCount;
+
+ ALOGV("avg. latency = %lld ms (max %lld ms)",
+ avgLatencyUs / 1000ll,
+ mLatencyMaxUs / 1000ll);
+
+ sp<AMessage> params = new AMessage;
+ params->setInt64("avgLatencyUs", avgLatencyUs);
+ params->setInt64("maxLatencyUs", mLatencyMaxUs);
+ mMediaReceiver->informSender(0 /* trackIndex */, params);
+ }
+
+ mLatencyCount = 0;
+ mLatencySumUs = 0ll;
+ mLatencyMaxUs = 0ll;
+
+ msg->post(kReportLatenessEveryUs);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void WifiDisplaySink::dumpDelay(size_t trackIndex, int64_t timeUs) {
+ int64_t delayMs = (ALooper::GetNowUs() - timeUs) / 1000ll;
+
+ if (delayMs > mMaxDelayMs) {
+ mMaxDelayMs = delayMs;
+ }
+
+ static const int64_t kMinDelayMs = 0;
+ static const int64_t kMaxDelayMs = 300;
+
+ const char *kPattern = "########################################";
+ size_t kPatternSize = strlen(kPattern);
+
+ int n = (kPatternSize * (delayMs - kMinDelayMs))
+ / (kMaxDelayMs - kMinDelayMs);
+
+ if (n < 0) {
+ n = 0;
+ } else if ((size_t)n > kPatternSize) {
+ n = kPatternSize;
+ }
+
+ ALOGI("[%lld]: (%4lld ms / %4lld ms) %s",
+ timeUs / 1000,
+ delayMs,
+ mMaxDelayMs,
+ kPattern + kPatternSize - n);
+}
+
+void WifiDisplaySink::onMediaReceiverNotify(const sp<AMessage> &msg) {
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case MediaReceiver::kWhatInitDone:
+ {
+ status_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ ALOGI("MediaReceiver initialization completed w/ err %d", err);
+ break;
+ }
+
+ case MediaReceiver::kWhatError:
+ {
+ status_t err;
+ CHECK(msg->findInt32("err", &err));
+
+ ALOGE("MediaReceiver signaled error %d", err);
+ break;
+ }
+
+ case MediaReceiver::kWhatAccessUnit:
+ {
+ if (mRenderer == NULL) {
+ mRenderer = new DirectRenderer(mSurfaceTex);
+ looper()->registerHandler(mRenderer);
+ }
+
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("accessUnit", &accessUnit));
+
+ int64_t timeUs;
+ CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+
+ if (!mTimeOffsetValid && !(mFlags & FLAG_SPECIAL_MODE)) {
+ mTimeOffsetUs = timeUs - ALooper::GetNowUs();
+ mTimeOffsetValid = true;
+ }
+
+ CHECK(mTimeOffsetValid);
+
+ // We are the timesync _client_,
+ // client time = server time - time offset.
+ timeUs -= mTimeOffsetUs;
+
+ size_t trackIndex;
+ CHECK(msg->findSize("trackIndex", &trackIndex));
+
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t delayUs = nowUs - timeUs;
+
+ mLatencySumUs += delayUs;
+ if (mLatencyCount == 0 || delayUs > mLatencyMaxUs) {
+ mLatencyMaxUs = delayUs;
+ }
+ ++mLatencyCount;
+
+ // dumpDelay(trackIndex, timeUs);
+
+ timeUs += 220000ll; // Assume 220 ms of latency
+ accessUnit->meta()->setInt64("timeUs", timeUs);
+
+ sp<AMessage> format;
+ if (msg->findMessage("format", &format)) {
+ mRenderer->setFormat(trackIndex, format);
+ }
+
+ mRenderer->queueAccessUnit(trackIndex, accessUnit);
+ break;
+ }
+
+ case MediaReceiver::kWhatPacketLost:
+ {
+#if 0
+ if (!mIDRFrameRequestPending) {
+ ALOGI("requesting IDR frame");
+
+ sendIDRFrameRequest(mSessionID);
+ }
+#endif
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void WifiDisplaySink::registerResponseHandler(
+ int32_t sessionID, int32_t cseq, HandleRTSPResponseFunc func) {
+ ResponseID id;
+ id.mSessionID = sessionID;
+ id.mCSeq = cseq;
+ mResponseHandlers.add(id, func);
+}
+
+status_t WifiDisplaySink::sendM2(int32_t sessionID) {
+ AString request = "OPTIONS * RTSP/1.0\r\n";
+ AppendCommonResponse(&request, mNextCSeq);
+
+ request.append(
+ "Require: org.wfa.wfd1.0\r\n"
+ "\r\n");
+
+ status_t err =
+ mNetSession->sendRequest(sessionID, request.c_str(), request.size());
+
+ if (err != OK) {
+ return err;
+ }
+
+ registerResponseHandler(
+ sessionID, mNextCSeq, &WifiDisplaySink::onReceiveM2Response);
+
+ ++mNextCSeq;
+
+ return OK;
+}
+
+status_t WifiDisplaySink::onReceiveM2Response(
+ int32_t sessionID, const sp<ParsedMessage> &msg) {
+ int32_t statusCode;
+ if (!msg->getStatusCode(&statusCode)) {
+ return ERROR_MALFORMED;
+ }
+
+ if (statusCode != 200) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ return OK;
+}
+
+status_t WifiDisplaySink::onReceiveSetupResponse(
+ int32_t sessionID, const sp<ParsedMessage> &msg) {
+ int32_t statusCode;
+ if (!msg->getStatusCode(&statusCode)) {
+ return ERROR_MALFORMED;
+ }
+
+ if (statusCode != 200) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ if (!msg->findString("session", &mPlaybackSessionID)) {
+ return ERROR_MALFORMED;
+ }
+
+ if (!ParsedMessage::GetInt32Attribute(
+ mPlaybackSessionID.c_str(),
+ "timeout",
+ &mPlaybackSessionTimeoutSecs)) {
+ mPlaybackSessionTimeoutSecs = -1;
+ }
+
+ ssize_t colonPos = mPlaybackSessionID.find(";");
+ if (colonPos >= 0) {
+ // Strip any options from the returned session id.
+ mPlaybackSessionID.erase(
+ colonPos, mPlaybackSessionID.size() - colonPos);
+ }
+
+ status_t err = configureTransport(msg);
+
+ if (err != OK) {
+ return err;
+ }
+
+ mState = PAUSED;
+
+ return sendPlay(
+ sessionID,
+ "rtsp://x.x.x.x:x/wfd1.0/streamid=0");
+}
+
+status_t WifiDisplaySink::configureTransport(const sp<ParsedMessage> &msg) {
+ if (mUsingTCPTransport && !(mFlags & FLAG_SPECIAL_MODE)) {
+ // In "special" mode we still use a UDP RTCP back-channel that
+ // needs connecting.
+ return OK;
+ }
+
+ AString transport;
+ if (!msg->findString("transport", &transport)) {
+ ALOGE("Missing 'transport' field in SETUP response.");
+ return ERROR_MALFORMED;
+ }
+
+ AString sourceHost;
+ if (!ParsedMessage::GetAttribute(
+ transport.c_str(), "source", &sourceHost)) {
+ sourceHost = mRTSPHost;
+ }
+
+ AString serverPortStr;
+ if (!ParsedMessage::GetAttribute(
+ transport.c_str(), "server_port", &serverPortStr)) {
+ ALOGE("Missing 'server_port' in Transport field.");
+ return ERROR_MALFORMED;
+ }
+
+ int rtpPort, rtcpPort;
+ if (sscanf(serverPortStr.c_str(), "%d-%d", &rtpPort, &rtcpPort) != 2
+ || rtpPort <= 0 || rtpPort > 65535
+ || rtcpPort <=0 || rtcpPort > 65535
+ || rtcpPort != rtpPort + 1) {
+ ALOGE("Invalid server_port description '%s'.",
+ serverPortStr.c_str());
+
+ return ERROR_MALFORMED;
+ }
+
+ if (rtpPort & 1) {
+ ALOGW("Server picked an odd numbered RTP port.");
+ }
+
+ return mMediaReceiver->connectTrack(
+ 0 /* trackIndex */, sourceHost.c_str(), rtpPort, rtcpPort);
+}
+
+status_t WifiDisplaySink::onReceivePlayResponse(
+ int32_t sessionID, const sp<ParsedMessage> &msg) {
+ int32_t statusCode;
+ if (!msg->getStatusCode(&statusCode)) {
+ return ERROR_MALFORMED;
+ }
+
+ if (statusCode != 200) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ mState = PLAYING;
+
+ (new AMessage(kWhatReportLateness, id()))->post(kReportLatenessEveryUs);
+
+ return OK;
+}
+
+status_t WifiDisplaySink::onReceiveIDRFrameRequestResponse(
+ int32_t sessionID, const sp<ParsedMessage> &msg) {
+ CHECK(mIDRFrameRequestPending);
+ mIDRFrameRequestPending = false;
+
+ return OK;
+}
+
+void WifiDisplaySink::onReceiveClientData(const sp<AMessage> &msg) {
+ int32_t sessionID;
+ CHECK(msg->findInt32("sessionID", &sessionID));
+
+ sp<RefBase> obj;
+ CHECK(msg->findObject("data", &obj));
+
+ sp<ParsedMessage> data =
+ static_cast<ParsedMessage *>(obj.get());
+
+ ALOGV("session %d received '%s'",
+ sessionID, data->debugString().c_str());
+
+ AString method;
+ AString uri;
+ data->getRequestField(0, &method);
+
+ int32_t cseq;
+ if (!data->findInt32("cseq", &cseq)) {
+ sendErrorResponse(sessionID, "400 Bad Request", -1 /* cseq */);
+ return;
+ }
+
+ if (method.startsWith("RTSP/")) {
+ // This is a response.
+
+ ResponseID id;
+ id.mSessionID = sessionID;
+ id.mCSeq = cseq;
+
+ ssize_t index = mResponseHandlers.indexOfKey(id);
+
+ if (index < 0) {
+ ALOGW("Received unsolicited server response, cseq %d", cseq);
+ return;
+ }
+
+ HandleRTSPResponseFunc func = mResponseHandlers.valueAt(index);
+ mResponseHandlers.removeItemsAt(index);
+
+ status_t err = (this->*func)(sessionID, data);
+ CHECK_EQ(err, (status_t)OK);
+ } else {
+ AString version;
+ data->getRequestField(2, &version);
+ if (!(version == AString("RTSP/1.0"))) {
+ sendErrorResponse(sessionID, "505 RTSP Version not supported", cseq);
+ return;
+ }
+
+ if (method == "OPTIONS") {
+ onOptionsRequest(sessionID, cseq, data);
+ } else if (method == "GET_PARAMETER") {
+ onGetParameterRequest(sessionID, cseq, data);
+ } else if (method == "SET_PARAMETER") {
+ onSetParameterRequest(sessionID, cseq, data);
+ } else {
+ sendErrorResponse(sessionID, "405 Method Not Allowed", cseq);
+ }
+ }
+}
+
+void WifiDisplaySink::onOptionsRequest(
+ int32_t sessionID,
+ int32_t cseq,
+ const sp<ParsedMessage> &data) {
+ AString response = "RTSP/1.0 200 OK\r\n";
+ AppendCommonResponse(&response, cseq);
+ response.append("Public: org.wfa.wfd1.0, GET_PARAMETER, SET_PARAMETER\r\n");
+ response.append("\r\n");
+
+ status_t err = mNetSession->sendRequest(sessionID, response.c_str());
+ CHECK_EQ(err, (status_t)OK);
+
+ err = sendM2(sessionID);
+ CHECK_EQ(err, (status_t)OK);
+}
+
+void WifiDisplaySink::onGetParameterRequest(
+ int32_t sessionID,
+ int32_t cseq,
+ const sp<ParsedMessage> &data) {
+ AString body;
+
+ if (mState == CONNECTED) {
+ mUsingTCPTransport = false;
+ mUsingTCPInterleaving = false;
+
+ char val[PROPERTY_VALUE_MAX];
+ if (property_get("media.wfd-sink.tcp-mode", val, NULL)) {
+ if (!strcasecmp("true", val) || !strcmp("1", val)) {
+ ALOGI("Using TCP unicast transport.");
+ mUsingTCPTransport = true;
+ mUsingTCPInterleaving = false;
+ } else if (!strcasecmp("interleaved", val)) {
+ ALOGI("Using TCP interleaved transport.");
+ mUsingTCPTransport = true;
+ mUsingTCPInterleaving = true;
+ }
+ } else if (mFlags & FLAG_SPECIAL_MODE) {
+ mUsingTCPTransport = true;
+ }
+
+ body = "wfd_video_formats: ";
+ body.append(mSinkSupportedVideoFormats.getFormatSpec());
+
+ body.append(
+ "\r\nwfd_audio_codecs: AAC 0000000F 00\r\n"
+ "wfd_client_rtp_ports: RTP/AVP/");
+
+ if (mUsingTCPTransport) {
+ body.append("TCP;");
+ if (mUsingTCPInterleaving) {
+ body.append("interleaved");
+ } else {
+ body.append("unicast 19000 0");
+ }
+ } else {
+ body.append("UDP;unicast 19000 0");
+ }
+
+ body.append(" mode=play\r\n");
+ }
+
+ AString response = "RTSP/1.0 200 OK\r\n";
+ AppendCommonResponse(&response, cseq);
+ response.append("Content-Type: text/parameters\r\n");
+ response.append(StringPrintf("Content-Length: %d\r\n", body.size()));
+ response.append("\r\n");
+ response.append(body);
+
+ status_t err = mNetSession->sendRequest(sessionID, response.c_str());
+ CHECK_EQ(err, (status_t)OK);
+}
+
+status_t WifiDisplaySink::sendSetup(int32_t sessionID, const char *uri) {
+ sp<AMessage> notify = new AMessage(kWhatMediaReceiverNotify, id());
+
+ mMediaReceiverLooper = new ALooper;
+ mMediaReceiverLooper->setName("media_receiver");
+
+ mMediaReceiverLooper->start(
+ false /* runOnCallingThread */,
+ false /* canCallJava */,
+ PRIORITY_AUDIO);
+
+ mMediaReceiver = new MediaReceiver(mNetSession, notify);
+ mMediaReceiverLooper->registerHandler(mMediaReceiver);
+
+ RTPReceiver::TransportMode rtpMode = RTPReceiver::TRANSPORT_UDP;
+ if (mUsingTCPTransport) {
+ if (mUsingTCPInterleaving) {
+ rtpMode = RTPReceiver::TRANSPORT_TCP_INTERLEAVED;
+ } else {
+ rtpMode = RTPReceiver::TRANSPORT_TCP;
+ }
+ }
+
+ int32_t localRTPPort;
+ status_t err = mMediaReceiver->addTrack(
+ rtpMode, RTPReceiver::TRANSPORT_UDP /* rtcpMode */, &localRTPPort);
+
+ if (err == OK) {
+ err = mMediaReceiver->initAsync(MediaReceiver::MODE_TRANSPORT_STREAM);
+ }
+
+ if (err != OK) {
+ mMediaReceiverLooper->unregisterHandler(mMediaReceiver->id());
+ mMediaReceiver.clear();
+
+ mMediaReceiverLooper->stop();
+ mMediaReceiverLooper.clear();
+
+ return err;
+ }
+
+ AString request = StringPrintf("SETUP %s RTSP/1.0\r\n", uri);
+
+ AppendCommonResponse(&request, mNextCSeq);
+
+ if (rtpMode == RTPReceiver::TRANSPORT_TCP_INTERLEAVED) {
+ request.append("Transport: RTP/AVP/TCP;interleaved=0-1\r\n");
+ } else if (rtpMode == RTPReceiver::TRANSPORT_TCP) {
+ if (mFlags & FLAG_SPECIAL_MODE) {
+ // This isn't quite true, since the RTP connection is through TCP
+ // and the RTCP connection through UDP...
+ request.append(
+ StringPrintf(
+ "Transport: RTP/AVP/TCP;unicast;client_port=%d-%d\r\n",
+ localRTPPort, localRTPPort + 1));
+ } else {
+ request.append(
+ StringPrintf(
+ "Transport: RTP/AVP/TCP;unicast;client_port=%d\r\n",
+ localRTPPort));
+ }
+ } else {
+ request.append(
+ StringPrintf(
+ "Transport: RTP/AVP/UDP;unicast;client_port=%d-%d\r\n",
+ localRTPPort,
+ localRTPPort + 1));
+ }
+
+ request.append("\r\n");
+
+ ALOGV("request = '%s'", request.c_str());
+
+ err = mNetSession->sendRequest(sessionID, request.c_str(), request.size());
+
+ if (err != OK) {
+ return err;
+ }
+
+ registerResponseHandler(
+ sessionID, mNextCSeq, &WifiDisplaySink::onReceiveSetupResponse);
+
+ ++mNextCSeq;
+
+ return OK;
+}
+
+status_t WifiDisplaySink::sendPlay(int32_t sessionID, const char *uri) {
+ AString request = StringPrintf("PLAY %s RTSP/1.0\r\n", uri);
+
+ AppendCommonResponse(&request, mNextCSeq);
+
+ request.append(StringPrintf("Session: %s\r\n", mPlaybackSessionID.c_str()));
+ request.append("\r\n");
+
+ status_t err =
+ mNetSession->sendRequest(sessionID, request.c_str(), request.size());
+
+ if (err != OK) {
+ return err;
+ }
+
+ registerResponseHandler(
+ sessionID, mNextCSeq, &WifiDisplaySink::onReceivePlayResponse);
+
+ ++mNextCSeq;
+
+ return OK;
+}
+
+status_t WifiDisplaySink::sendIDRFrameRequest(int32_t sessionID) {
+ CHECK(!mIDRFrameRequestPending);
+
+ AString request = "SET_PARAMETER rtsp://localhost/wfd1.0 RTSP/1.0\r\n";
+
+ AppendCommonResponse(&request, mNextCSeq);
+
+ AString content = "wfd_idr_request\r\n";
+
+ request.append(StringPrintf("Session: %s\r\n", mPlaybackSessionID.c_str()));
+ request.append(StringPrintf("Content-Length: %d\r\n", content.size()));
+ request.append("\r\n");
+ request.append(content);
+
+ status_t err =
+ mNetSession->sendRequest(sessionID, request.c_str(), request.size());
+
+ if (err != OK) {
+ return err;
+ }
+
+ registerResponseHandler(
+ sessionID,
+ mNextCSeq,
+ &WifiDisplaySink::onReceiveIDRFrameRequestResponse);
+
+ ++mNextCSeq;
+
+ mIDRFrameRequestPending = true;
+
+ return OK;
+}
+
+void WifiDisplaySink::onSetParameterRequest(
+ int32_t sessionID,
+ int32_t cseq,
+ const sp<ParsedMessage> &data) {
+ const char *content = data->getContent();
+
+ if (strstr(content, "wfd_trigger_method: SETUP\r\n") != NULL) {
+ if ((mFlags & FLAG_SPECIAL_MODE) && !mTimeOffsetValid) {
+ mSetupDeferred = true;
+ } else {
+ status_t err =
+ sendSetup(
+ sessionID,
+ "rtsp://x.x.x.x:x/wfd1.0/streamid=0");
+
+ CHECK_EQ(err, (status_t)OK);
+ }
+ }
+
+ AString response = "RTSP/1.0 200 OK\r\n";
+ AppendCommonResponse(&response, cseq);
+ response.append("\r\n");
+
+ status_t err = mNetSession->sendRequest(sessionID, response.c_str());
+ CHECK_EQ(err, (status_t)OK);
+}
+
+void WifiDisplaySink::sendErrorResponse(
+ int32_t sessionID,
+ const char *errorDetail,
+ int32_t cseq) {
+ AString response;
+ response.append("RTSP/1.0 ");
+ response.append(errorDetail);
+ response.append("\r\n");
+
+ AppendCommonResponse(&response, cseq);
+
+ response.append("\r\n");
+
+ status_t err = mNetSession->sendRequest(sessionID, response.c_str());
+ CHECK_EQ(err, (status_t)OK);
+}
+
+// static
+void WifiDisplaySink::AppendCommonResponse(AString *response, int32_t cseq) {
+ time_t now = time(NULL);
+ struct tm *now2 = gmtime(&now);
+ char buf[128];
+ strftime(buf, sizeof(buf), "%a, %d %b %Y %H:%M:%S %z", now2);
+
+ response->append("Date: ");
+ response->append(buf);
+ response->append("\r\n");
+
+ response->append(StringPrintf("User-Agent: %s\r\n", sUserAgent.c_str()));
+
+ if (cseq >= 0) {
+ response->append(StringPrintf("CSeq: %d\r\n", cseq));
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/wifi-display/sink/WifiDisplaySink.h b/media/libstagefright/wifi-display/sink/WifiDisplaySink.h
new file mode 100644
index 0000000..adb9d89
--- /dev/null
+++ b/media/libstagefright/wifi-display/sink/WifiDisplaySink.h
@@ -0,0 +1,196 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef WIFI_DISPLAY_SINK_H_
+
+#define WIFI_DISPLAY_SINK_H_
+
+#include "ANetworkSession.h"
+
+#include "VideoFormats.h"
+
+#include <gui/Surface.h>
+#include <media/stagefright/foundation/AHandler.h>
+
+namespace android {
+
+struct AMessage;
+struct DirectRenderer;
+struct MediaReceiver;
+struct ParsedMessage;
+struct TimeSyncer;
+
+// Represents the RTSP client acting as a wifi display sink.
+// Connects to a wifi display source and renders the incoming
+// transport stream using a MediaPlayer instance.
+struct WifiDisplaySink : public AHandler {
+ enum {
+ kWhatDisconnected,
+ };
+
+ enum Flags {
+ FLAG_SPECIAL_MODE = 1,
+ };
+
+ // If no notification message is specified (notify == NULL)
+ // the sink will stop its looper() once the session ends,
+ // otherwise it will post an appropriate notification but leave
+ // the looper() running.
+ WifiDisplaySink(
+ uint32_t flags,
+ const sp<ANetworkSession> &netSession,
+ const sp<IGraphicBufferProducer> &bufferProducer = NULL,
+ const sp<AMessage> ¬ify = NULL);
+
+ void start(const char *sourceHost, int32_t sourcePort);
+ void start(const char *uri);
+
+protected:
+ virtual ~WifiDisplaySink();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum State {
+ UNDEFINED,
+ CONNECTING,
+ CONNECTED,
+ PAUSED,
+ PLAYING,
+ };
+
+ enum {
+ kWhatStart,
+ kWhatRTSPNotify,
+ kWhatStop,
+ kWhatMediaReceiverNotify,
+ kWhatTimeSyncerNotify,
+ kWhatReportLateness,
+ };
+
+ struct ResponseID {
+ int32_t mSessionID;
+ int32_t mCSeq;
+
+ bool operator<(const ResponseID &other) const {
+ return mSessionID < other.mSessionID
+ || (mSessionID == other.mSessionID
+ && mCSeq < other.mCSeq);
+ }
+ };
+
+ typedef status_t (WifiDisplaySink::*HandleRTSPResponseFunc)(
+ int32_t sessionID, const sp<ParsedMessage> &msg);
+
+ static const int64_t kReportLatenessEveryUs = 1000000ll;
+
+ static const AString sUserAgent;
+
+ State mState;
+ uint32_t mFlags;
+ VideoFormats mSinkSupportedVideoFormats;
+ sp<ANetworkSession> mNetSession;
+ sp<IGraphicBufferProducer> mSurfaceTex;
+ sp<AMessage> mNotify;
+ sp<TimeSyncer> mTimeSyncer;
+ bool mUsingTCPTransport;
+ bool mUsingTCPInterleaving;
+ AString mRTSPHost;
+ int32_t mSessionID;
+
+ int32_t mNextCSeq;
+
+ KeyedVector<ResponseID, HandleRTSPResponseFunc> mResponseHandlers;
+
+ sp<ALooper> mMediaReceiverLooper;
+ sp<MediaReceiver> mMediaReceiver;
+ sp<DirectRenderer> mRenderer;
+
+ AString mPlaybackSessionID;
+ int32_t mPlaybackSessionTimeoutSecs;
+
+ bool mIDRFrameRequestPending;
+
+ int64_t mTimeOffsetUs;
+ bool mTimeOffsetValid;
+
+ bool mSetupDeferred;
+
+ size_t mLatencyCount;
+ int64_t mLatencySumUs;
+ int64_t mLatencyMaxUs;
+
+ int64_t mMaxDelayMs;
+
+ status_t sendM2(int32_t sessionID);
+ status_t sendSetup(int32_t sessionID, const char *uri);
+ status_t sendPlay(int32_t sessionID, const char *uri);
+ status_t sendIDRFrameRequest(int32_t sessionID);
+
+ status_t onReceiveM2Response(
+ int32_t sessionID, const sp<ParsedMessage> &msg);
+
+ status_t onReceiveSetupResponse(
+ int32_t sessionID, const sp<ParsedMessage> &msg);
+
+ status_t configureTransport(const sp<ParsedMessage> &msg);
+
+ status_t onReceivePlayResponse(
+ int32_t sessionID, const sp<ParsedMessage> &msg);
+
+ status_t onReceiveIDRFrameRequestResponse(
+ int32_t sessionID, const sp<ParsedMessage> &msg);
+
+ void registerResponseHandler(
+ int32_t sessionID, int32_t cseq, HandleRTSPResponseFunc func);
+
+ void onReceiveClientData(const sp<AMessage> &msg);
+
+ void onOptionsRequest(
+ int32_t sessionID,
+ int32_t cseq,
+ const sp<ParsedMessage> &data);
+
+ void onGetParameterRequest(
+ int32_t sessionID,
+ int32_t cseq,
+ const sp<ParsedMessage> &data);
+
+ void onSetParameterRequest(
+ int32_t sessionID,
+ int32_t cseq,
+ const sp<ParsedMessage> &data);
+
+ void onMediaReceiverNotify(const sp<AMessage> &msg);
+
+ void sendErrorResponse(
+ int32_t sessionID,
+ const char *errorDetail,
+ int32_t cseq);
+
+ static void AppendCommonResponse(AString *response, int32_t cseq);
+
+ bool ParseURL(
+ const char *url, AString *host, int32_t *port, AString *path,
+ AString *user, AString *pass);
+
+ void dumpDelay(size_t trackIndex, int64_t timeUs);
+
+ DISALLOW_EVIL_CONSTRUCTORS(WifiDisplaySink);
+};
+
+} // namespace android
+
+#endif // WIFI_DISPLAY_SINK_H_
diff --git a/media/libstagefright/wifi-display/source/Converter.cpp b/media/libstagefright/wifi-display/source/Converter.cpp
index 5344623..0214520 100644
--- a/media/libstagefright/wifi-display/source/Converter.cpp
+++ b/media/libstagefright/wifi-display/source/Converter.cpp
@@ -40,14 +40,13 @@
Converter::Converter(
const sp<AMessage> ¬ify,
const sp<ALooper> &codecLooper,
- const sp<AMessage> &format,
- bool usePCMAudio)
+ const sp<AMessage> &outputFormat)
: mInitCheck(NO_INIT),
mNotify(notify),
mCodecLooper(codecLooper),
- mInputFormat(format),
+ mOutputFormat(outputFormat),
mIsVideo(false),
- mIsPCMAudio(usePCMAudio),
+ mIsPCMAudio(false),
mNeedToManuallyPrependSPSPPS(false),
mDoMoreWorkPending(false)
#if ENABLE_SILENCE_DETECTION
@@ -58,14 +57,14 @@
,mNumFramesToDrop(0)
{
AString mime;
- CHECK(mInputFormat->findString("mime", &mime));
+ CHECK(mOutputFormat->findString("mime", &mime));
if (!strncasecmp("video/", mime.c_str(), 6)) {
mIsVideo = true;
+ } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_RAW, mime.c_str())) {
+ mIsPCMAudio = true;
}
- CHECK(!usePCMAudio || !mIsVideo);
-
mInitCheck = initEncoder();
if (mInitCheck != OK) {
@@ -152,23 +151,10 @@
}
status_t Converter::initEncoder() {
- AString inputMIME;
- CHECK(mInputFormat->findString("mime", &inputMIME));
-
AString outputMIME;
- bool isAudio = false;
- if (!strcasecmp(inputMIME.c_str(), MEDIA_MIMETYPE_AUDIO_RAW)) {
- if (mIsPCMAudio) {
- outputMIME = MEDIA_MIMETYPE_AUDIO_RAW;
- } else {
- outputMIME = MEDIA_MIMETYPE_AUDIO_AAC;
- }
- isAudio = true;
- } else if (!strcasecmp(inputMIME.c_str(), MEDIA_MIMETYPE_VIDEO_RAW)) {
- outputMIME = MEDIA_MIMETYPE_VIDEO_AVC;
- } else {
- TRESPASS();
- }
+ CHECK(mOutputFormat->findString("mime", &outputMIME));
+
+ bool isAudio = !strncasecmp(outputMIME.c_str(), "audio/", 6);
if (!mIsPCMAudio) {
mEncoder = MediaCodec::CreateByType(
@@ -179,14 +165,10 @@
}
}
- mOutputFormat = mInputFormat->dup();
-
if (mIsPCMAudio) {
return OK;
}
- mOutputFormat->setString("mime", outputMIME.c_str());
-
int32_t audioBitrate = GetInt32Property("media.wfd.audio-bitrate", 128000);
int32_t videoBitrate = GetInt32Property("media.wfd.video-bitrate", 5000000);
mPrevVideoBitrate = videoBitrate;
@@ -427,7 +409,7 @@
releaseEncoder();
AString mime;
- CHECK(mInputFormat->findString("mime", &mime));
+ CHECK(mOutputFormat->findString("mime", &mime));
ALOGI("encoder (%s) shut down.", mime.c_str());
break;
}
@@ -438,6 +420,17 @@
break;
}
+ case kWhatReleaseOutputBuffer:
+ {
+ if (mEncoder != NULL) {
+ size_t bufferIndex;
+ CHECK(msg->findInt32("bufferIndex", (int32_t*)&bufferIndex));
+ CHECK(bufferIndex < mEncoderOutputBuffers.size());
+ mEncoder->releaseOutputBuffer(bufferIndex);
+ }
+ break;
+ }
+
default:
TRESPASS();
}
@@ -645,6 +638,7 @@
size_t size;
int64_t timeUs;
uint32_t flags;
+ native_handle_t* handle = NULL;
err = mEncoder->dequeueOutputBuffer(
&bufferIndex, &offset, &size, &timeUs, &flags);
@@ -667,18 +661,52 @@
notify->setInt32("what", kWhatEOS);
notify->post();
} else {
- sp<ABuffer> buffer = new ABuffer(size);
+#if 0
+ if (mIsVideo) {
+ int32_t videoBitrate = GetInt32Property(
+ "media.wfd.video-bitrate", 5000000);
+
+ setVideoBitrate(videoBitrate);
+ }
+#endif
+
+ sp<ABuffer> buffer;
+ sp<ABuffer> outbuf = mEncoderOutputBuffers.itemAt(bufferIndex);
+
+ if (outbuf->meta()->findPointer("handle", (void**)&handle) &&
+ handle != NULL) {
+ int32_t rangeLength, rangeOffset;
+ CHECK(outbuf->meta()->findInt32("rangeOffset", &rangeOffset));
+ CHECK(outbuf->meta()->findInt32("rangeLength", &rangeLength));
+ outbuf->meta()->setPointer("handle", NULL);
+
+ // MediaSender will post the following message when HDCP
+ // is done, to release the output buffer back to encoder.
+ sp<AMessage> notify(new AMessage(
+ kWhatReleaseOutputBuffer, id()));
+ notify->setInt32("bufferIndex", bufferIndex);
+
+ buffer = new ABuffer(
+ rangeLength > (int32_t)size ? rangeLength : size);
+ buffer->meta()->setPointer("handle", handle);
+ buffer->meta()->setInt32("rangeOffset", rangeOffset);
+ buffer->meta()->setInt32("rangeLength", rangeLength);
+ buffer->meta()->setMessage("notify", notify);
+ } else {
+ buffer = new ABuffer(size);
+ }
+
buffer->meta()->setInt64("timeUs", timeUs);
ALOGV("[%s] time %lld us (%.2f secs)",
mIsVideo ? "video" : "audio", timeUs, timeUs / 1E6);
- memcpy(buffer->data(),
- mEncoderOutputBuffers.itemAt(bufferIndex)->base() + offset,
- size);
+ memcpy(buffer->data(), outbuf->base() + offset, size);
if (flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) {
- mOutputFormat->setBuffer("csd-0", buffer);
+ if (!handle) {
+ mOutputFormat->setBuffer("csd-0", buffer);
+ }
} else {
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatAccessUnit);
@@ -687,7 +715,9 @@
}
}
- mEncoder->releaseOutputBuffer(bufferIndex);
+ if (!handle) {
+ mEncoder->releaseOutputBuffer(bufferIndex);
+ }
if (flags & MediaCodec::BUFFER_FLAG_EOS) {
break;
diff --git a/media/libstagefright/wifi-display/source/Converter.h b/media/libstagefright/wifi-display/source/Converter.h
index ba297c4..76c8b19 100644
--- a/media/libstagefright/wifi-display/source/Converter.h
+++ b/media/libstagefright/wifi-display/source/Converter.h
@@ -33,11 +33,9 @@
// media access unit of a different format.
// Right now this'll convert raw video into H.264 and raw audio into AAC.
struct Converter : public AHandler {
- Converter(
- const sp<AMessage> ¬ify,
- const sp<ALooper> &codecLooper,
- const sp<AMessage> &format,
- bool usePCMAudio);
+ Converter(const sp<AMessage> ¬ify,
+ const sp<ALooper> &codecLooper,
+ const sp<AMessage> &outputFormat);
status_t initCheck() const;
@@ -66,6 +64,7 @@
kWhatMediaPullerNotify,
kWhatEncoderActivity,
kWhatDropAFrame,
+ kWhatReleaseOutputBuffer,
};
void shutdownAsync();
@@ -83,10 +82,9 @@
status_t mInitCheck;
sp<AMessage> mNotify;
sp<ALooper> mCodecLooper;
- sp<AMessage> mInputFormat;
+ sp<AMessage> mOutputFormat;
bool mIsVideo;
bool mIsPCMAudio;
- sp<AMessage> mOutputFormat;
bool mNeedToManuallyPrependSPSPPS;
sp<MediaCodec> mEncoder;
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
index 3d7b865..a15fbac 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
@@ -378,7 +378,9 @@
bool usePCMAudio,
bool enableVideo,
VideoFormats::ResolutionType videoResolutionType,
- size_t videoResolutionIndex) {
+ size_t videoResolutionIndex,
+ VideoFormats::ProfileType videoProfileType,
+ VideoFormats::LevelType videoLevelType) {
sp<AMessage> notify = new AMessage(kWhatMediaSenderNotify, id());
mMediaSender = new MediaSender(mNetSession, notify);
looper()->registerHandler(mMediaSender);
@@ -390,7 +392,9 @@
usePCMAudio,
enableVideo,
videoResolutionType,
- videoResolutionIndex);
+ videoResolutionIndex,
+ videoProfileType,
+ videoLevelType);
if (err == OK) {
err = mMediaSender->initAsync(
@@ -559,6 +563,8 @@
converter->dropAFrame();
}
}
+ } else if (what == MediaSender::kWhatInformSender) {
+ onSinkFeedback(msg);
} else {
TRESPASS();
}
@@ -654,6 +660,89 @@
}
}
+void WifiDisplaySource::PlaybackSession::onSinkFeedback(const sp<AMessage> &msg) {
+ int64_t avgLatencyUs;
+ CHECK(msg->findInt64("avgLatencyUs", &avgLatencyUs));
+
+ int64_t maxLatencyUs;
+ CHECK(msg->findInt64("maxLatencyUs", &maxLatencyUs));
+
+ ALOGI("sink reports avg. latency of %lld ms (max %lld ms)",
+ avgLatencyUs / 1000ll,
+ maxLatencyUs / 1000ll);
+
+ if (mVideoTrackIndex >= 0) {
+ const sp<Track> &videoTrack = mTracks.valueFor(mVideoTrackIndex);
+ sp<Converter> converter = videoTrack->converter();
+
+ if (converter != NULL) {
+ int32_t videoBitrate =
+ Converter::GetInt32Property("media.wfd.video-bitrate", -1);
+
+ char val[PROPERTY_VALUE_MAX];
+ if (videoBitrate < 0
+ && property_get("media.wfd.video-bitrate", val, NULL)
+ && !strcasecmp("adaptive", val)) {
+ videoBitrate = converter->getVideoBitrate();
+
+ if (avgLatencyUs > 300000ll) {
+ videoBitrate *= 0.6;
+ } else if (avgLatencyUs < 100000ll) {
+ videoBitrate *= 1.1;
+ }
+ }
+
+ if (videoBitrate > 0) {
+ if (videoBitrate < 500000) {
+ videoBitrate = 500000;
+ } else if (videoBitrate > 10000000) {
+ videoBitrate = 10000000;
+ }
+
+ if (videoBitrate != converter->getVideoBitrate()) {
+ ALOGI("setting video bitrate to %d bps", videoBitrate);
+
+ converter->setVideoBitrate(videoBitrate);
+ }
+ }
+ }
+
+ sp<RepeaterSource> repeaterSource = videoTrack->repeaterSource();
+ if (repeaterSource != NULL) {
+ double rateHz =
+ Converter::GetInt32Property(
+ "media.wfd.video-framerate", -1);
+
+ char val[PROPERTY_VALUE_MAX];
+ if (rateHz < 0.0
+ && property_get("media.wfd.video-framerate", val, NULL)
+ && !strcasecmp("adaptive", val)) {
+ rateHz = repeaterSource->getFrameRate();
+
+ if (avgLatencyUs > 300000ll) {
+ rateHz *= 0.9;
+ } else if (avgLatencyUs < 200000ll) {
+ rateHz *= 1.1;
+ }
+ }
+
+ if (rateHz > 0) {
+ if (rateHz < 5.0) {
+ rateHz = 5.0;
+ } else if (rateHz > 30.0) {
+ rateHz = 30.0;
+ }
+
+ if (rateHz != repeaterSource->getFrameRate()) {
+ ALOGI("setting frame rate to %.2f Hz", rateHz);
+
+ repeaterSource->setFrameRate(rateHz);
+ }
+ }
+ }
+ }
+}
+
status_t WifiDisplaySource::PlaybackSession::setupMediaPacketizer(
bool enableAudio, bool enableVideo) {
DataSource::RegisterDefaultSniffers();
@@ -785,7 +874,9 @@
bool usePCMAudio,
bool enableVideo,
VideoFormats::ResolutionType videoResolutionType,
- size_t videoResolutionIndex) {
+ size_t videoResolutionIndex,
+ VideoFormats::ProfileType videoProfileType,
+ VideoFormats::LevelType videoLevelType) {
CHECK(enableAudio || enableVideo);
if (!mMediaPath.empty()) {
@@ -794,7 +885,8 @@
if (enableVideo) {
status_t err = addVideoSource(
- videoResolutionType, videoResolutionIndex);
+ videoResolutionType, videoResolutionIndex, videoProfileType,
+ videoLevelType);
if (err != OK) {
return err;
@@ -810,9 +902,13 @@
status_t WifiDisplaySource::PlaybackSession::addSource(
bool isVideo, const sp<MediaSource> &source, bool isRepeaterSource,
- bool usePCMAudio, size_t *numInputBuffers) {
+ bool usePCMAudio, unsigned profileIdc, unsigned levelIdc,
+ unsigned constraintSet, size_t *numInputBuffers) {
CHECK(!usePCMAudio || !isVideo);
CHECK(!isRepeaterSource || isVideo);
+ CHECK(!profileIdc || isVideo);
+ CHECK(!levelIdc || isVideo);
+ CHECK(!constraintSet || isVideo);
sp<ALooper> pullLooper = new ALooper;
pullLooper->setName("pull_looper");
@@ -841,17 +937,25 @@
CHECK_EQ(err, (status_t)OK);
if (isVideo) {
+ format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC);
format->setInt32("store-metadata-in-buffers", true);
-
+ format->setInt32("store-metadata-in-buffers-output", (mHDCP != NULL));
format->setInt32(
"color-format", OMX_COLOR_FormatAndroidOpaque);
+ format->setInt32("profile-idc", profileIdc);
+ format->setInt32("level-idc", levelIdc);
+ format->setInt32("constraint-set", constraintSet);
+ } else {
+ format->setString(
+ "mime",
+ usePCMAudio
+ ? MEDIA_MIMETYPE_AUDIO_RAW : MEDIA_MIMETYPE_AUDIO_AAC);
}
notify = new AMessage(kWhatConverterNotify, id());
notify->setSize("trackIndex", trackIndex);
- sp<Converter> converter =
- new Converter(notify, codecLooper, format, usePCMAudio);
+ sp<Converter> converter = new Converter(notify, codecLooper, format);
err = converter->initCheck();
if (err != OK) {
@@ -905,7 +1009,9 @@
status_t WifiDisplaySource::PlaybackSession::addVideoSource(
VideoFormats::ResolutionType videoResolutionType,
- size_t videoResolutionIndex) {
+ size_t videoResolutionIndex,
+ VideoFormats::ProfileType videoProfileType,
+ VideoFormats::LevelType videoLevelType) {
size_t width, height, framesPerSecond;
bool interlaced;
CHECK(VideoFormats::GetConfiguration(
@@ -916,6 +1022,14 @@
&framesPerSecond,
&interlaced));
+ unsigned profileIdc, levelIdc, constraintSet;
+ CHECK(VideoFormats::GetProfileLevel(
+ videoProfileType,
+ videoLevelType,
+ &profileIdc,
+ &levelIdc,
+ &constraintSet));
+
sp<SurfaceMediaSource> source = new SurfaceMediaSource(width, height);
source->setUseAbsoluteTimestamps();
@@ -926,7 +1040,8 @@
size_t numInputBuffers;
status_t err = addSource(
true /* isVideo */, videoSource, true /* isRepeaterSource */,
- false /* usePCMAudio */, &numInputBuffers);
+ false /* usePCMAudio */, profileIdc, levelIdc, constraintSet,
+ &numInputBuffers);
if (err != OK) {
return err;
@@ -949,7 +1064,8 @@
if (audioSource->initCheck() == OK) {
return addSource(
false /* isVideo */, audioSource, false /* isRepeaterSource */,
- usePCMAudio, NULL /* numInputBuffers */);
+ usePCMAudio, 0 /* profileIdc */, 0 /* levelIdc */,
+ 0 /* constraintSet */, NULL /* numInputBuffers */);
}
ALOGW("Unable to instantiate audio source");
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.h b/media/libstagefright/wifi-display/source/PlaybackSession.h
index 39086a1..5c8ee94 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.h
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.h
@@ -53,7 +53,9 @@
bool usePCMAudio,
bool enableVideo,
VideoFormats::ResolutionType videoResolutionType,
- size_t videoResolutionIndex);
+ size_t videoResolutionIndex,
+ VideoFormats::ProfileType videoProfileType,
+ VideoFormats::LevelType videoLevelType);
void destroyAsync();
@@ -130,18 +132,25 @@
bool usePCMAudio,
bool enableVideo,
VideoFormats::ResolutionType videoResolutionType,
- size_t videoResolutionIndex);
+ size_t videoResolutionIndex,
+ VideoFormats::ProfileType videoProfileType,
+ VideoFormats::LevelType videoLevelType);
status_t addSource(
bool isVideo,
const sp<MediaSource> &source,
bool isRepeaterSource,
bool usePCMAudio,
+ unsigned profileIdc,
+ unsigned levelIdc,
+ unsigned contraintSet,
size_t *numInputBuffers);
status_t addVideoSource(
VideoFormats::ResolutionType videoResolutionType,
- size_t videoResolutionIndex);
+ size_t videoResolutionIndex,
+ VideoFormats::ProfileType videoProfileType,
+ VideoFormats::LevelType videoLevelType);
status_t addAudioSource(bool usePCMAudio);
diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index 2c4a373..c674700 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -261,12 +261,24 @@
data[0] = 40; // descriptor_tag
data[1] = 4; // descriptor_length
- CHECK_GE(mCSD.size(), 1u);
- const sp<ABuffer> &sps = mCSD.itemAt(0);
- CHECK(!memcmp("\x00\x00\x00\x01", sps->data(), 4));
- CHECK_GE(sps->size(), 7u);
- // profile_idc, constraint_set*, level_idc
- memcpy(&data[2], sps->data() + 4, 3);
+ if (mCSD.size() > 0) {
+ CHECK_GE(mCSD.size(), 1u);
+ const sp<ABuffer> &sps = mCSD.itemAt(0);
+ CHECK(!memcmp("\x00\x00\x00\x01", sps->data(), 4));
+ CHECK_GE(sps->size(), 7u);
+ // profile_idc, constraint_set*, level_idc
+ memcpy(&data[2], sps->data() + 4, 3);
+ } else {
+ int32_t profileIdc, levelIdc, constraintSet;
+ CHECK(mFormat->findInt32("profile-idc", &profileIdc));
+ CHECK(mFormat->findInt32("level-idc", &levelIdc));
+ CHECK(mFormat->findInt32("constraint-set", &constraintSet));
+ CHECK_GE(profileIdc, 0u);
+ CHECK_GE(levelIdc, 0u);
+ data[2] = profileIdc; // profile_idc
+ data[3] = constraintSet; // constraint_set*
+ data[4] = levelIdc; // level_idc
+ }
// AVC_still_present=0, AVC_24_hour_picture_flag=0, reserved
data[5] = 0x3f;
diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
index 22dd0b1..b421b35 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
@@ -23,6 +23,7 @@
#include "Parameters.h"
#include "ParsedMessage.h"
#include "rtp/RTPSender.h"
+#include "TimeSyncer.h"
#include <binder/IServiceManager.h>
#include <gui/IGraphicBufferProducer.h>
@@ -73,6 +74,12 @@
mSupportedSourceVideoFormats.setNativeResolution(
VideoFormats::RESOLUTION_CEA, 5); // 1280x720 p30
+
+ // Enable all resolutions up to 1280x720p30
+ mSupportedSourceVideoFormats.enableResolutionUpto(
+ VideoFormats::RESOLUTION_CEA, 5,
+ VideoFormats::PROFILE_CHP, // Constrained High Profile
+ VideoFormats::LEVEL_32); // Level 3.2
}
WifiDisplaySource::~WifiDisplaySource() {
@@ -164,6 +171,14 @@
} else {
err = -EINVAL;
}
+ }
+
+ if (err == OK) {
+ sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id());
+ mTimeSyncer = new TimeSyncer(mNetSession, notify);
+ looper()->registerHandler(mTimeSyncer);
+
+ mTimeSyncer->startServer(8123);
mState = AWAITING_CLIENT_CONNECTION;
}
@@ -539,6 +554,11 @@
break;
}
+ case kWhatTimeSyncerNotify:
+ {
+ break;
+ }
+
default:
TRESPASS();
}
@@ -617,6 +637,9 @@
chosenVideoFormat.disableAll();
chosenVideoFormat.setNativeResolution(
mChosenVideoResolutionType, mChosenVideoResolutionIndex);
+ chosenVideoFormat.setProfileLevel(
+ mChosenVideoResolutionType, mChosenVideoResolutionIndex,
+ mChosenVideoProfile, mChosenVideoLevel);
body.append(chosenVideoFormat.getFormatSpec(true /* forM4Message */));
body.append("\r\n");
@@ -729,6 +752,8 @@
++mNextCSeq;
+ scheduleKeepAlive(sessionID);
+
return OK;
}
@@ -845,7 +870,9 @@
mSupportedSinkVideoFormats,
mSupportedSourceVideoFormats,
&mChosenVideoResolutionType,
- &mChosenVideoResolutionIndex)) {
+ &mChosenVideoResolutionIndex,
+ &mChosenVideoProfile,
+ &mChosenVideoLevel)) {
ALOGE("Sink and source share no commonly supported video "
"formats.");
@@ -864,6 +891,9 @@
ALOGI("Picked video resolution %u x %u %c%u",
width, height, interlaced ? 'i' : 'p', framesPerSecond);
+
+ ALOGI("Picked AVC profile %d, level %d",
+ mChosenVideoProfile, mChosenVideoLevel);
} else {
ALOGI("Sink doesn't support video at all.");
}
@@ -994,8 +1024,6 @@
if (mClientInfo.mPlaybackSession != NULL) {
mClientInfo.mPlaybackSession->updateLiveness();
-
- scheduleKeepAlive(sessionID);
}
return OK;
@@ -1257,7 +1285,9 @@
mUsingPCMAudio,
mSinkSupportsVideo,
mChosenVideoResolutionType,
- mChosenVideoResolutionIndex);
+ mChosenVideoResolutionIndex,
+ mChosenVideoProfile,
+ mChosenVideoLevel);
if (err != OK) {
looper()->unregisterHandler(playbackSession->id());
@@ -1340,7 +1370,9 @@
return ERROR_MALFORMED;
}
- if (mState != AWAITING_CLIENT_PLAY) {
+ if (mState != AWAITING_CLIENT_PLAY
+ && mState != PAUSED_TO_PLAYING
+ && mState != PAUSED) {
ALOGW("Received PLAY request but we're in state %d", mState);
sendErrorResponse(
@@ -1367,7 +1399,7 @@
return err;
}
- if (mState == PAUSED_TO_PLAYING) {
+ if (mState == PAUSED_TO_PLAYING || mPlaybackSessionEstablished) {
mState = PLAYING;
return OK;
}
@@ -1401,7 +1433,7 @@
ALOGI("Received PAUSE request.");
- if (mState != PLAYING_TO_PAUSED) {
+ if (mState != PLAYING_TO_PAUSED && mState != PLAYING) {
return INVALID_OPERATION;
}
diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.h b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
index 44d3e4d..64186fc 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.h
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
@@ -30,6 +30,7 @@
struct IHDCP;
struct IRemoteDisplayClient;
struct ParsedMessage;
+struct TimeSyncer;
// Represents the RTSP server acting as a wifi display source.
// Manages incoming connections, sets up Playback sessions as necessary.
@@ -82,6 +83,7 @@
kWhatHDCPNotify,
kWhatFinishStop2,
kWhatTeardownTriggerTimedOut,
+ kWhatTimeSyncerNotify,
};
struct ResponseID {
@@ -118,6 +120,7 @@
sp<ANetworkSession> mNetSession;
sp<IRemoteDisplayClient> mClient;
AString mMediaPath;
+ sp<TimeSyncer> mTimeSyncer;
struct in_addr mInterfaceAddr;
int32_t mSessionID;
@@ -131,6 +134,8 @@
VideoFormats::ResolutionType mChosenVideoResolutionType;
size_t mChosenVideoResolutionIndex;
+ VideoFormats::ProfileType mChosenVideoProfile;
+ VideoFormats::LevelType mChosenVideoLevel;
bool mSinkSupportsAudio;
diff --git a/media/libstagefright/wifi-display/udptest.cpp b/media/libstagefright/wifi-display/udptest.cpp
new file mode 100644
index 0000000..111846d
--- /dev/null
+++ b/media/libstagefright/wifi-display/udptest.cpp
@@ -0,0 +1,116 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NEBUG 0
+#define LOG_TAG "udptest"
+#include <utils/Log.h>
+
+#include "ANetworkSession.h"
+#include "TimeSyncer.h"
+
+#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+namespace android {
+
+} // namespace android
+
+static void usage(const char *me) {
+ fprintf(stderr,
+ "usage: %s -c host[:port]\tconnect to test server\n"
+ " -l \tcreate a test server\n",
+ me);
+}
+
+int main(int argc, char **argv) {
+ using namespace android;
+
+ ProcessState::self()->startThreadPool();
+
+ int32_t localPort = -1;
+ int32_t connectToPort = -1;
+ AString connectToHost;
+
+ int res;
+ while ((res = getopt(argc, argv, "hc:l:")) >= 0) {
+ switch (res) {
+ case 'c':
+ {
+ const char *colonPos = strrchr(optarg, ':');
+
+ if (colonPos == NULL) {
+ connectToHost = optarg;
+ connectToPort = 49152;
+ } else {
+ connectToHost.setTo(optarg, colonPos - optarg);
+
+ char *end;
+ connectToPort = strtol(colonPos + 1, &end, 10);
+
+ if (*end != '\0' || end == colonPos + 1
+ || connectToPort < 1 || connectToPort > 65535) {
+ fprintf(stderr, "Illegal port specified.\n");
+ exit(1);
+ }
+ }
+ break;
+ }
+
+ case 'l':
+ {
+ char *end;
+ localPort = strtol(optarg, &end, 10);
+
+ if (*end != '\0' || end == optarg
+ || localPort < 1 || localPort > 65535) {
+ fprintf(stderr, "Illegal port specified.\n");
+ exit(1);
+ }
+ break;
+ }
+
+ case '?':
+ case 'h':
+ usage(argv[0]);
+ exit(1);
+ }
+ }
+
+ if (localPort < 0 && connectToPort < 0) {
+ fprintf(stderr,
+ "You need to select either client or server mode.\n");
+ exit(1);
+ }
+
+ sp<ANetworkSession> netSession = new ANetworkSession;
+ netSession->start();
+
+ sp<ALooper> looper = new ALooper;
+
+ sp<TimeSyncer> handler = new TimeSyncer(netSession, NULL /* notify */);
+ looper->registerHandler(handler);
+
+ if (localPort >= 0) {
+ handler->startServer(localPort);
+ } else {
+ handler->startClient(connectToHost.c_str(), connectToPort);
+ }
+
+ looper->start(true /* runOnCallingThread */);
+
+ return 0;
+}
+
diff --git a/media/libstagefright/wifi-display/wfd.cpp b/media/libstagefright/wifi-display/wfd.cpp
index c947765..9fee4d0 100644
--- a/media/libstagefright/wifi-display/wfd.cpp
+++ b/media/libstagefright/wifi-display/wfd.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "wfd"
#include <utils/Log.h>
+#include "sink/WifiDisplaySink.h"
#include "source/WifiDisplaySource.h"
#include <binder/ProcessState.h>
@@ -38,8 +39,12 @@
static void usage(const char *me) {
fprintf(stderr,
"usage:\n"
- " %s -l iface[:port]\tcreate a wifi display source\n"
- " -f(ilename) \tstream media\n",
+ " %s -c host[:port]\tconnect to wifi source\n"
+ " -u uri \tconnect to an rtsp uri\n"
+ " -l ip[:port] \tlisten on the specified port "
+ " -f(ilename) \tstream media "
+ "(create a sink)\n"
+ " -s(pecial) \trun in 'special' mode\n",
me);
}
@@ -209,14 +214,48 @@
DataSource::RegisterDefaultSniffers();
+ AString connectToHost;
+ int32_t connectToPort = -1;
+ AString uri;
+
AString listenOnAddr;
int32_t listenOnPort = -1;
AString path;
+ bool specialMode = false;
+
int res;
- while ((res = getopt(argc, argv, "hl:f:")) >= 0) {
+ while ((res = getopt(argc, argv, "hc:l:u:f:s")) >= 0) {
switch (res) {
+ case 'c':
+ {
+ const char *colonPos = strrchr(optarg, ':');
+
+ if (colonPos == NULL) {
+ connectToHost = optarg;
+ connectToPort = WifiDisplaySource::kWifiDisplayDefaultPort;
+ } else {
+ connectToHost.setTo(optarg, colonPos - optarg);
+
+ char *end;
+ connectToPort = strtol(colonPos + 1, &end, 10);
+
+ if (*end != '\0' || end == colonPos + 1
+ || connectToPort < 1 || connectToPort > 65535) {
+ fprintf(stderr, "Illegal port specified.\n");
+ exit(1);
+ }
+ }
+ break;
+ }
+
+ case 'u':
+ {
+ uri = optarg;
+ break;
+ }
+
case 'f':
{
path = optarg;
@@ -245,6 +284,12 @@
break;
}
+ case 's':
+ {
+ specialMode = true;
+ break;
+ }
+
case '?':
case 'h':
default:
@@ -253,6 +298,13 @@
}
}
+ if (connectToPort >= 0 && listenOnPort >= 0) {
+ fprintf(stderr,
+ "You can connect to a source or create one, "
+ "but not both at the same time.\n");
+ exit(1);
+ }
+
if (listenOnPort >= 0) {
if (path.empty()) {
createSource(listenOnAddr, listenOnPort);
@@ -263,7 +315,72 @@
exit(0);
}
- usage(argv[0]);
+ if (connectToPort < 0 && uri.empty()) {
+ fprintf(stderr,
+ "You need to select either source host or uri.\n");
+
+ exit(1);
+ }
+
+ if (connectToPort >= 0 && !uri.empty()) {
+ fprintf(stderr,
+ "You need to either connect to a wfd host or an rtsp url, "
+ "not both.\n");
+ exit(1);
+ }
+
+ sp<SurfaceComposerClient> composerClient = new SurfaceComposerClient;
+ CHECK_EQ(composerClient->initCheck(), (status_t)OK);
+
+ sp<IBinder> display(SurfaceComposerClient::getBuiltInDisplay(
+ ISurfaceComposer::eDisplayIdMain));
+ DisplayInfo info;
+ SurfaceComposerClient::getDisplayInfo(display, &info);
+ ssize_t displayWidth = info.w;
+ ssize_t displayHeight = info.h;
+
+ ALOGV("display is %d x %d\n", displayWidth, displayHeight);
+
+ sp<SurfaceControl> control =
+ composerClient->createSurface(
+ String8("A Surface"),
+ displayWidth,
+ displayHeight,
+ PIXEL_FORMAT_RGB_565,
+ 0);
+
+ CHECK(control != NULL);
+ CHECK(control->isValid());
+
+ SurfaceComposerClient::openGlobalTransaction();
+ CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK);
+ CHECK_EQ(control->show(), (status_t)OK);
+ SurfaceComposerClient::closeGlobalTransaction();
+
+ sp<Surface> surface = control->getSurface();
+ CHECK(surface != NULL);
+
+ sp<ANetworkSession> session = new ANetworkSession;
+ session->start();
+
+ sp<ALooper> looper = new ALooper;
+
+ sp<WifiDisplaySink> sink = new WifiDisplaySink(
+ specialMode ? WifiDisplaySink::FLAG_SPECIAL_MODE : 0 /* flags */,
+ session,
+ surface->getIGraphicBufferProducer());
+
+ looper->registerHandler(sink);
+
+ if (connectToPort >= 0) {
+ sink->start(connectToHost.c_str(), connectToPort);
+ } else {
+ sink->start(uri.c_str());
+ }
+
+ looper->start(true /* runOnCallingThread */);
+
+ composerClient->dispose();
return 0;
}
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 061a079..714854e 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -81,6 +81,8 @@
LOCAL_CFLAGS += -DANDROID_SMP=0
endif
+LOCAL_CFLAGS += -fvisibility=hidden
+
include $(BUILD_SHARED_LIBRARY)
#
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 87eb6aa..a6edb77 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1453,10 +1453,18 @@
}
mPlaybackThreads.add(id, thread);
- if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate;
- if (pFormat != NULL) *pFormat = config.format;
- if (pChannelMask != NULL) *pChannelMask = config.channel_mask;
- if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = config.sample_rate;
+ }
+ if (pFormat != NULL) {
+ *pFormat = config.format;
+ }
+ if (pChannelMask != NULL) {
+ *pChannelMask = config.channel_mask;
+ }
+ if (pLatencyMs != NULL) {
+ *pLatencyMs = thread->latency();
+ }
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
@@ -1698,9 +1706,15 @@
);
mRecordThreads.add(id, thread);
ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
- if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
- if (pFormat != NULL) *pFormat = config.format;
- if (pChannelMask != NULL) *pChannelMask = reqChannels;
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = reqSamplingRate;
+ }
+ if (pFormat != NULL) {
+ *pFormat = config.format;
+ }
+ if (pChannelMask != NULL) {
+ *pChannelMask = reqChannels;
+ }
// notify client processes of the new input creation
thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index b0efef6..05dbab1 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -24,6 +24,8 @@
#include <common_time/cc_helper.h>
+#include <cutils/compiler.h>
+
#include <media/IAudioFlinger.h>
#include <media/IAudioFlingerClient.h>
#include <media/IAudioTrack.h>
@@ -54,6 +56,7 @@
#include <powermanager/IPowerManager.h>
#include <media/nbaio/NBLog.h>
+#include <private/media/AudioTrackShared.h>
namespace android {
@@ -89,7 +92,7 @@
{
friend class BinderService<AudioFlinger>; // for AudioFlinger()
public:
- static const char* getServiceName() { return "media.audio_flinger"; }
+ static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; }
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -278,7 +281,7 @@
bool btNrecIsOff() const { return mBtNrecIsOff; }
- AudioFlinger();
+ AudioFlinger() ANDROID_API;
virtual ~AudioFlinger();
// call in any IAudioFlinger method that accesses mPrimaryHardwareDev
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 35cf368..53238fa 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -19,6 +19,7 @@
#include <cutils/misc.h>
#include <cutils/config_utils.h>
+#include <cutils/compiler.h>
#include <utils/String8.h>
#include <utils/Vector.h>
#include <utils/SortedVector.h>
@@ -44,7 +45,7 @@
public:
// for BinderService
- static const char *getServiceName() { return "media.audio_policy"; }
+ static const char *getServiceName() ANDROID_API { return "media.audio_policy"; }
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -137,7 +138,7 @@
virtual status_t setVoiceVolume(float volume, int delayMs = 0);
private:
- AudioPolicyService();
+ AudioPolicyService() ANDROID_API;
virtual ~AudioPolicyService();
status_t dumpInternals(int fd);
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 2b8694f..29dc5b6 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -19,13 +19,14 @@
#include <stdint.h>
#include <sys/types.h>
+#include <cutils/compiler.h>
#include <media/AudioBufferProvider.h>
namespace android {
// ----------------------------------------------------------------------------
-class AudioResampler {
+class ANDROID_API AudioResampler {
public:
// Determines quality of SRC.
// LOW_QUALITY: linear interpolator (1st order)
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index a749d7a..b1286d3 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -46,6 +46,8 @@
void destroy();
int name() const { return mName; }
+ virtual uint32_t sampleRate() const;
+
audio_stream_type_t streamType() const {
return mStreamType;
}
@@ -139,6 +141,7 @@
// 'volatile' means accessed without lock or
// barrier, but is read/written atomically
bool mIsInvalid; // non-resettable latch, set by invalidate()
+ AudioTrackServerProxy* mAudioTrackServerProxy;
}; // end of Track
class TimedTrack : public Track {
@@ -255,10 +258,6 @@
private:
- enum {
- NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value
- };
-
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer,
uint32_t waitTimeMs);
void clearBufferQueue();
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 6c0d1d3..ffe3e9f 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -57,4 +57,5 @@
// releaseBuffer() not overridden
bool mOverflow; // overflow on most recent attempt to fill client buffer
+ AudioRecordServerProxy* mAudioRecordServerProxy;
};
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 97f66f4..0773534 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -139,7 +139,7 @@
// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
// N-buffering, so AudioFlinger could allocate the right amount of memory.
// See the client's minBufCount and mNotificationFramesAct calculations for details.
-static const int kFastTrackMultiplier = 2;
+static const int kFastTrackMultiplier = 1;
// ----------------------------------------------------------------------------
@@ -495,7 +495,8 @@
sp<IBinder> binder = new BBinder();
status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
- String16(mName));
+ String16(mName),
+ String16("media"));
if (status == NO_ERROR) {
mWakeLockToken = binder;
}
@@ -1326,7 +1327,7 @@
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
- track->mFillingUpStatus = Track::FS_FILLING;
+ track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
track->mResetDone = false;
track->mPresentationCompleteFrames = 0;
mActiveTracks.add(track);
@@ -1713,7 +1714,7 @@
void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
{
- ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
+ ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
this, streamType, mTracks.size());
Mutex::Autolock _l(mLock);
@@ -2595,24 +2596,35 @@
// app does not call stop() and relies on underrun to stop:
// hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
// during last round
+ size_t desiredFrames;
+ if (t->sampleRate() == mSampleRate) {
+ desiredFrames = mNormalFrameCount;
+ } else {
+ // +1 for rounding and +1 for additional sample needed for interpolation
+ desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
+ // add frames already consumed but not yet released by the resampler
+ // because cblk->framesReady() will include these frames
+ desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+ // the minimum track buffer size is normally twice the number of frames necessary
+ // to fill one buffer and the resampler should not leave more than one buffer worth
+ // of unreleased frames after each pass, but just in case...
+ ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
+ }
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
- if (t->sampleRate() == mSampleRate) {
- minFrames = mNormalFrameCount;
- } else {
- // +1 for rounding and +1 for additional sample needed for interpolation
- minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
- // add frames already consumed but not yet released by the resampler
- // because cblk->framesReady() will include these frames
- minFrames += mAudioMixer->getUnreleasedFrames(track->name());
- // the minimum track buffer size is normally twice the number of frames necessary
- // to fill one buffer and the resampler should not leave more than one buffer worth
- // of unreleased frames after each pass, but just in case...
- ALOG_ASSERT(minFrames <= cblk->frameCount_);
- }
+ minFrames = desiredFrames;
}
- if ((track->framesReady() >= minFrames) && track->isReady() &&
+ // It's not safe to call framesReady() for a static buffer track, so assume it's ready
+ size_t framesReady;
+ if (track->sharedBuffer() == 0) {
+ framesReady = track->framesReady();
+ } else if (track->isStopped()) {
+ framesReady = 0;
+ } else {
+ framesReady = 1;
+ }
+ if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
@@ -2663,7 +2675,7 @@
// read original volumes with volume control
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
- ServerProxy *proxy = track->mServerProxy;
+ AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
uint32_t vlr = proxy->getVolumeLR();
vl = vlr & 0xFFFF;
vr = vlr >> 16;
@@ -2736,7 +2748,7 @@
AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * 2;
- uint32_t reqSampleRate = track->mServerProxy->getSampleRate();
+ uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
if (reqSampleRate == 0) {
reqSampleRate = mSampleRate;
} else if (reqSampleRate > maxSampleRate) {
@@ -2767,6 +2779,13 @@
mixerStatus = MIXER_TRACKS_READY;
}
} else {
+ // only implemented for normal tracks, not fast tracks
+ if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
+ // we missed desiredFrames whatever the actual number of frames missing was
+ cblk->u.mStreaming.mUnderrunFrames += desiredFrames;
+ // FIXME also wake futex so that underrun is noticed more quickly
+ (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags);
+ }
// clear effect chain input buffer if an active track underruns to avoid sending
// previous audio buffer again to effects
chain = getEffectChain_l(track->sessionId());
@@ -3169,7 +3188,7 @@
} else {
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
- uint32_t vlr = track->mServerProxy->getVolumeLR();
+ uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR();
float v_clamped = v * (vlr & 0xFFFF);
if (v_clamped > MAX_GAIN) {
v_clamped = MAX_GAIN;
@@ -3695,7 +3714,8 @@
}
buffer.frameCount = mFrameCount;
- if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ status_t status = mActiveTrack->getNextBuffer(&buffer);
+ if (CC_LIKELY(status == NO_ERROR)) {
readOnce = true;
size_t framesOut = buffer.frameCount;
if (mResampler == NULL) {
@@ -3969,6 +3989,7 @@
ALOGV("Record started OK");
return status;
}
+
startError:
AudioSystem::stopInput(mId);
clearSyncStartEvent();
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index fac7071..55d96fa 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -74,7 +74,7 @@
audio_channel_mask_t channelMask() const { return mChannelMask; }
- uint32_t sampleRate() const; // FIXME inline after cblk sr moved
+ virtual uint32_t sampleRate() const { return mSampleRate; }
// Return a pointer to the start of a contiguous slice of the track buffer.
// Parameter 'offset' is the requested start position, expressed in
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 5ac3129..bfc197c 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -98,7 +98,7 @@
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
- size_t bufferSize = frameCount * mFrameSize;
+ size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
if (sharedBuffer == 0) {
size += bufferSize;
}
@@ -124,22 +124,16 @@
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount_ = frameCount;
-// uncomment the following lines to quickly test 32-bit wraparound
-// mCblk->user = 0xffff0000;
-// mCblk->server = 0xffff0000;
-// mCblk->userBase = 0xffff0000;
-// mCblk->serverBase = 0xffff0000;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
- // Force underrun condition to avoid false underrun callback until first data is
- // written to buffer (other flags are cleared)
- mCblk->flags = CBLK_UNDERRUN;
} else {
mBuffer = sharedBuffer->pointer();
+#if 0
+ mCblk->flags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic
+#endif
}
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
- mServerProxy = new ServerProxy(mCblk, mBuffer, frameCount, mFrameSize, isOut);
#ifdef TEE_SINK
if (mTeeSinkTrackEnabled) {
@@ -199,51 +193,17 @@
}
#endif
- buffer->raw = NULL;
- mStepCount = buffer->frameCount;
- // FIXME See note at getNextBuffer()
- (void) step(); // ignore return value of step()
+ ServerProxy::Buffer buf;
+ buf.mFrameCount = buffer->frameCount;
+ buf.mRaw = buffer->raw;
buffer->frameCount = 0;
-}
-
-bool AudioFlinger::ThreadBase::TrackBase::step() {
- bool result = mServerProxy->step(mStepCount);
- if (!result) {
- ALOGV("stepServer failed acquiring cblk mutex");
- mStepServerFailed = true;
- }
- return result;
+ buffer->raw = NULL;
+ mServerProxy->releaseBuffer(&buf);
}
void AudioFlinger::ThreadBase::TrackBase::reset() {
- audio_track_cblk_t* cblk = this->cblk();
-
- cblk->user = 0;
- cblk->server = 0;
- cblk->userBase = 0;
- cblk->serverBase = 0;
- mStepServerFailed = false;
ALOGV("TrackBase::reset");
-}
-
-uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
- return mServerProxy->getSampleRate();
-}
-
-void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
- audio_track_cblk_t* cblk = this->cblk();
- int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
- int8_t *bufferEnd = bufferStart + frames * mFrameSize;
-
- // Check validity of returned pointer in case the track control block would have been corrupted.
- ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
- "TrackBase::getBuffer buffer out of range:\n"
- " start: %p, end %p , mBuffer %p mBufferEnd %p\n"
- " server %u, serverBase %u, user %u, userBase %u, frameSize %u",
- bufferStart, bufferEnd, mBuffer, mBufferEnd,
- cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
-
- return bufferStart;
+ // FIXME still needed?
}
status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
@@ -362,9 +322,18 @@
mFastIndex(-1),
mUnderrunCount(0),
mCachedVolume(1.0),
- mIsInvalid(false)
+ mIsInvalid(false),
+ mAudioTrackServerProxy(NULL)
{
if (mCblk != NULL) {
+ if (sharedBuffer == 0) {
+ mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
+ mFrameSize);
+ } else {
+ mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
+ mFrameSize);
+ }
+ mServerProxy = mAudioTrackServerProxy;
// to avoid leaking a track name, do not allocate one unless there is an mCblk
mName = thread->getTrackName_l(channelMask, sessionId);
mCblk->mName = mName;
@@ -374,6 +343,7 @@
}
// only allocate a fast track index if we were able to allocate a normal track name
if (flags & IAudioFlinger::TRACK_FAST) {
+ mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
int i = __builtin_ctz(thread->mFastTrackAvailMask);
ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
@@ -432,12 +402,12 @@
/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
{
result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S F SRate "
- "L dB R dB Server User Main buf Aux Buf Flags Underruns\n");
+ "L dB R dB Server Main buf Aux Buf Flags Underruns\n");
}
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
- uint32_t vlr = mServerProxy->getVolumeLR();
+ uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
if (isFastTrack()) {
sprintf(buffer, " F %2d", mFastIndex);
} else {
@@ -496,7 +466,7 @@
break;
}
snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %5u %5.2g %5.2g "
- "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
+ "0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
@@ -506,11 +476,10 @@
mFrameCount,
stateChar,
mFillingUpStatus,
- mServerProxy->getSampleRate(),
+ mAudioTrackServerProxy->getSampleRate(),
20.0 * log10((vlr & 0xFFFF) / 4096.0),
20.0 * log10((vlr >> 16) / 4096.0),
mCblk->server,
- mCblk->user,
(int)mMainBuffer,
(int)mAuxBuffer,
mCblk->flags,
@@ -518,53 +487,27 @@
nowInUnderrun);
}
+uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
+ return mAudioTrackServerProxy->getSampleRate();
+}
+
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
AudioBufferProvider::Buffer* buffer, int64_t pts)
{
- audio_track_cblk_t* cblk = this->cblk();
- uint32_t framesReady;
- uint32_t framesReq = buffer->frameCount;
-
- // Check if last stepServer failed, try to step now
- if (mStepServerFailed) {
- // FIXME When called by fast mixer, this takes a mutex with tryLock().
- // Since the fast mixer is higher priority than client callback thread,
- // it does not result in priority inversion for client.
- // But a non-blocking solution would be preferable to avoid
- // fast mixer being unable to tryLock(), and
- // to avoid the extra context switches if the client wakes up,
- // discovers the mutex is locked, then has to wait for fast mixer to unlock.
- if (!step()) goto getNextBuffer_exit;
- ALOGV("stepServer recovered");
- mStepServerFailed = false;
+ ServerProxy::Buffer buf;
+ size_t desiredFrames = buffer->frameCount;
+ buf.mFrameCount = desiredFrames;
+ status_t status = mServerProxy->obtainBuffer(&buf);
+ buffer->frameCount = buf.mFrameCount;
+ buffer->raw = buf.mRaw;
+ if (buf.mFrameCount == 0) {
+ // only implemented so far for normal tracks, not fast tracks
+ mCblk->u.mStreaming.mUnderrunFrames += desiredFrames;
+ // FIXME also wake futex so that underrun is noticed more quickly
+ (void) android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
}
-
- // FIXME Same as above
- framesReady = mServerProxy->framesReady();
-
- if (CC_LIKELY(framesReady)) {
- uint32_t s = cblk->server;
- uint32_t bufferEnd = cblk->serverBase + mFrameCount;
-
- bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
- if (framesReq > framesReady) {
- framesReq = framesReady;
- }
- if (framesReq > bufferEnd - s) {
- framesReq = bufferEnd - s;
- }
-
- buffer->raw = getBuffer(s, framesReq);
- buffer->frameCount = framesReq;
- return NO_ERROR;
- }
-
-getNextBuffer_exit:
- buffer->raw = NULL;
- buffer->frameCount = 0;
- ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
- return NOT_ENOUGH_DATA;
+ return status;
}
// Note that framesReady() takes a mutex on the control block using tryLock().
@@ -576,7 +519,7 @@
// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
- return mServerProxy->framesReady();
+ return mAudioTrackServerProxy->framesReady();
}
// Don't call for fast tracks; the framesReady() could result in priority inversion
@@ -732,7 +675,6 @@
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
- android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
mFillingUpStatus = FS_FILLING;
mResetDone = true;
if (mState == FLUSHED) {
@@ -833,7 +775,7 @@
{
// called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
- uint32_t vlr = mServerProxy->getVolumeLR();
+ uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
uint32_t vl = vlr & 0xFFFF;
uint32_t vr = vlr >> 16;
// track volumes come from shared memory, so can't be trusted and must be clamped
@@ -870,9 +812,12 @@
void AudioFlinger::PlaybackThread::Track::invalidate()
{
- // FIXME should use proxy
- android_atomic_or(CBLK_INVALID, &mCblk->flags);
- mCblk->cv.signal();
+ // FIXME should use proxy, and needs work
+ audio_track_cblk_t* cblk = mCblk;
+ android_atomic_or(CBLK_INVALID, &cblk->flags);
+ android_atomic_release_store(0x40000000, &cblk->mFutex);
+ // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
+ (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
mIsInvalid = true;
}
@@ -1418,6 +1363,8 @@
mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
mClientProxy->setSendLevel(0.0);
mClientProxy->setSampleRate(sampleRate);
+ mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
+ true /*clientInServer*/);
} else {
ALOGW("Error creating output track on thread %p", playbackThread);
}
@@ -1477,7 +1424,7 @@
memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else {
- ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
+ ALOGW("OutputTrack::write() %p no more buffers in queue", this);
}
}
}
@@ -1498,9 +1445,10 @@
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
- if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
- ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
- mThread.unsafe_get());
+ status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
+ if (status != NO_ERROR) {
+ ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
+ mThread.unsafe_get(), status);
outputBufferFull = true;
break;
}
@@ -1515,7 +1463,10 @@
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
- mClientProxy->stepUser(outFrames);
+ Proxy::Buffer buf;
+ buf.mFrameCount = outFrames;
+ buf.mRaw = NULL;
+ mClientProxy->releaseBuffer(&buf);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channelCount;
mOutBuffer.frameCount -= outFrames;
@@ -1559,8 +1510,10 @@
// If no more buffers are pending, fill output track buffer to make sure it is started
// by output mixer.
if (frames == 0 && mBufferQueue.size() == 0) {
- if (mCblk->user < mFrameCount) {
- frames = mFrameCount - mCblk->user;
+ // FIXME borken, replace by getting framesReady() from proxy
+ size_t user = 0; // was mCblk->user
+ if (user < mFrameCount) {
+ frames = mFrameCount - user;
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[frames * channelCount];
pInBuffer->frameCount = frames;
@@ -1578,46 +1531,17 @@
status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
- audio_track_cblk_t* cblk = mCblk;
- uint32_t framesReq = buffer->frameCount;
-
- ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
- buffer->frameCount = 0;
-
- size_t framesAvail;
- {
- Mutex::Autolock _l(cblk->lock);
-
- // read the server count again
- while (!(framesAvail = mClientProxy->framesAvailable_l())) {
- if (CC_UNLIKELY(!mActive)) {
- ALOGV("Not active and NO_MORE_BUFFERS");
- return NO_MORE_BUFFERS;
- }
- status_t result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
- if (result != NO_ERROR) {
- return NO_MORE_BUFFERS;
- }
- }
- }
-
- if (framesReq > framesAvail) {
- framesReq = framesAvail;
- }
-
- uint32_t u = cblk->user;
- uint32_t bufferEnd = cblk->userBase + mFrameCount;
-
- if (framesReq > bufferEnd - u) {
- framesReq = bufferEnd - u;
- }
-
- buffer->frameCount = framesReq;
- buffer->raw = mClientProxy->buffer(u);
- return NO_ERROR;
+ ClientProxy::Buffer buf;
+ buf.mFrameCount = buffer->frameCount;
+ struct timespec timeout;
+ timeout.tv_sec = waitTimeMs / 1000;
+ timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
+ status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
+ buffer->frameCount = buf.mFrameCount;
+ buffer->raw = buf.mRaw;
+ return status;
}
-
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
@@ -1688,6 +1612,11 @@
mOverflow(false)
{
ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+ if (mCblk != NULL) {
+ mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
+ mFrameSize);
+ mServerProxy = mAudioRecordServerProxy;
+ }
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
@@ -1699,42 +1628,16 @@
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts)
{
- audio_track_cblk_t* cblk = this->cblk();
- uint32_t framesAvail;
- uint32_t framesReq = buffer->frameCount;
-
- // Check if last stepServer failed, try to step now
- if (mStepServerFailed) {
- if (!step()) {
- goto getNextBuffer_exit;
- }
- ALOGV("stepServer recovered");
- mStepServerFailed = false;
+ ServerProxy::Buffer buf;
+ buf.mFrameCount = buffer->frameCount;
+ status_t status = mServerProxy->obtainBuffer(&buf);
+ buffer->frameCount = buf.mFrameCount;
+ buffer->raw = buf.mRaw;
+ if (buf.mFrameCount == 0) {
+ // FIXME also wake futex so that overrun is noticed more quickly
+ (void) android_atomic_or(CBLK_OVERRUN, &mCblk->flags);
}
-
- // FIXME lock is not actually held, so overrun is possible
- framesAvail = mServerProxy->framesAvailableIn_l();
-
- if (CC_LIKELY(framesAvail)) {
- uint32_t s = cblk->server;
- uint32_t bufferEnd = cblk->serverBase + mFrameCount;
-
- if (framesReq > framesAvail) {
- framesReq = framesAvail;
- }
- if (framesReq > bufferEnd - s) {
- framesReq = bufferEnd - s;
- }
-
- buffer->raw = getBuffer(s, framesReq);
- buffer->frameCount = framesReq;
- return NO_ERROR;
- }
-
-getNextBuffer_exit:
- buffer->raw = NULL;
- buffer->frameCount = 0;
- return NOT_ENOUGH_DATA;
+ return status;
}
status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
@@ -1790,12 +1693,12 @@
/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
{
- result.append(" Clien Fmt Chn mask Session Step S Serv User FrameCount\n");
+ result.append(" Clien Fmt Chn mask Session Step S Serv FrameCount\n");
}
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
- snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %08x %05d\n",
+ snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %05d\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
@@ -1803,7 +1706,6 @@
mStepCount,
mState,
mCblk->server,
- mCblk->user,
mFrameCount);
}
diff --git a/services/camera/libcameraservice/Camera2Client.cpp b/services/camera/libcameraservice/Camera2Client.cpp
index a1971e3..d1ab7eb 100644
--- a/services/camera/libcameraservice/Camera2Client.cpp
+++ b/services/camera/libcameraservice/Camera2Client.cpp
@@ -616,17 +616,31 @@
params.previewCallbackOneShot = true;
}
if (params.previewCallbackFlags != (uint32_t)flag) {
+
+ if (flag != CAMERA_FRAME_CALLBACK_FLAG_NOOP) {
+ // Disable any existing preview callback window when enabling
+ // preview callback flags
+ res = mCallbackProcessor->setCallbackWindow(NULL);
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to clear preview callback surface:"
+ " %s (%d)", __FUNCTION__, mCameraId, strerror(-res), res);
+ return;
+ }
+ params.previewCallbackSurface = false;
+ }
+
params.previewCallbackFlags = flag;
+
switch(params.state) {
- case Parameters::PREVIEW:
- res = startPreviewL(params, true);
- break;
- case Parameters::RECORD:
- case Parameters::VIDEO_SNAPSHOT:
- res = startRecordingL(params, true);
- break;
- default:
- break;
+ case Parameters::PREVIEW:
+ res = startPreviewL(params, true);
+ break;
+ case Parameters::RECORD:
+ case Parameters::VIDEO_SNAPSHOT:
+ res = startRecordingL(params, true);
+ break;
+ default:
+ break;
}
if (res != OK) {
ALOGE("%s: Camera %d: Unable to refresh request in state %s",
@@ -637,6 +651,59 @@
}
+status_t Camera2Client::setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer) {
+ ATRACE_CALL();
+ ALOGV("%s: E", __FUNCTION__);
+ Mutex::Autolock icl(mBinderSerializationLock);
+ status_t res;
+ if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
+
+ sp<ANativeWindow> window;
+ if (callbackProducer != 0) {
+ window = new Surface(callbackProducer);
+ }
+
+ res = mCallbackProcessor->setCallbackWindow(window);
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to set preview callback surface: %s (%d)",
+ __FUNCTION__, mCameraId, strerror(-res), res);
+ return res;
+ }
+
+ SharedParameters::Lock l(mParameters);
+
+ if (window != NULL) {
+ // Disable traditional callbacks when a valid callback target is given
+ l.mParameters.previewCallbackFlags = CAMERA_FRAME_CALLBACK_FLAG_NOOP;
+ l.mParameters.previewCallbackOneShot = false;
+ l.mParameters.previewCallbackSurface = true;
+ } else {
+ // Disable callback target if given a NULL interface.
+ l.mParameters.previewCallbackSurface = false;
+ }
+
+ switch(l.mParameters.state) {
+ case Parameters::PREVIEW:
+ res = startPreviewL(l.mParameters, true);
+ break;
+ case Parameters::RECORD:
+ case Parameters::VIDEO_SNAPSHOT:
+ res = startRecordingL(l.mParameters, true);
+ break;
+ default:
+ break;
+ }
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to refresh request in state %s",
+ __FUNCTION__, mCameraId,
+ Parameters::getStateName(l.mParameters.state));
+ }
+
+ return OK;
+}
+
+
status_t Camera2Client::startPreview() {
ATRACE_CALL();
ALOGV("%s: E", __FUNCTION__);
@@ -699,8 +766,10 @@
}
Vector<uint8_t> outputStreams;
- bool callbacksEnabled = params.previewCallbackFlags &
- CAMERA_FRAME_CALLBACK_FLAG_ENABLE_MASK;
+ bool callbacksEnabled = (params.previewCallbackFlags &
+ CAMERA_FRAME_CALLBACK_FLAG_ENABLE_MASK) ||
+ params.previewCallbackSurface;
+
if (callbacksEnabled) {
res = mCallbackProcessor->updateStream(params);
if (res != OK) {
@@ -909,8 +978,10 @@
}
Vector<uint8_t> outputStreams;
- bool callbacksEnabled = params.previewCallbackFlags &
- CAMERA_FRAME_CALLBACK_FLAG_ENABLE_MASK;
+ bool callbacksEnabled = (params.previewCallbackFlags &
+ CAMERA_FRAME_CALLBACK_FLAG_ENABLE_MASK) ||
+ params.previewCallbackSurface;
+
if (callbacksEnabled) {
res = mCallbackProcessor->updateStream(params);
if (res != OK) {
diff --git a/services/camera/libcameraservice/Camera2Client.h b/services/camera/libcameraservice/Camera2Client.h
index 8ab46b1..078e3a3 100644
--- a/services/camera/libcameraservice/Camera2Client.h
+++ b/services/camera/libcameraservice/Camera2Client.h
@@ -51,6 +51,9 @@
virtual status_t setPreviewTexture(
const sp<IGraphicBufferProducer>& bufferProducer);
virtual void setPreviewCallbackFlag(int flag);
+ virtual status_t setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer);
+
virtual status_t startPreview();
virtual void stopPreview();
virtual bool previewEnabled();
diff --git a/services/camera/libcameraservice/Camera3Device.cpp b/services/camera/libcameraservice/Camera3Device.cpp
index cc7802b..73bf30c 100644
--- a/services/camera/libcameraservice/Camera3Device.cpp
+++ b/services/camera/libcameraservice/Camera3Device.cpp
@@ -128,7 +128,10 @@
/** Initialize device with callback functions */
+ ATRACE_BEGIN("camera3->initialize");
res = device->ops->initialize(device, this);
+ ATRACE_END();
+
if (res != OK) {
SET_ERR_L("Unable to initialize HAL device: %s (%d)",
strerror(-res), res);
@@ -140,7 +143,9 @@
mVendorTagOps.get_camera_vendor_section_name = NULL;
+ ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops");
device->ops->get_metadata_vendor_tag_ops(device, &mVendorTagOps);
+ ATRACE_END();
if (mVendorTagOps.get_camera_vendor_section_name != NULL) {
res = set_camera_metadata_vendor_tag_ops(&mVendorTagOps);
@@ -736,7 +741,7 @@
status_t Camera3Device::createDefaultRequest(int templateId,
CameraMetadata *request) {
ATRACE_CALL();
- ALOGV("%s: E", __FUNCTION__);
+ ALOGV("%s: for template %d", __FUNCTION__, templateId);
Mutex::Autolock l(mLock);
switch (mStatus) {
@@ -756,8 +761,10 @@
}
const camera_metadata_t *rawRequest;
+ ATRACE_BEGIN("camera3->construct_default_request_settings");
rawRequest = mHal3Device->ops->construct_default_request_settings(
mHal3Device, templateId);
+ ATRACE_END();
if (rawRequest == NULL) {
SET_ERR_L("HAL is unable to construct default settings for template %d",
templateId);
@@ -1068,8 +1075,9 @@
// Do the HAL configuration; will potentially touch stream
// max_buffers, usage, priv fields.
-
+ ATRACE_BEGIN("camera3->configure_streams");
res = mHal3Device->ops->configure_streams(mHal3Device, &config);
+ ATRACE_END();
if (res != OK) {
SET_ERR_L("Unable to configure streams with HAL: %s (%d)",
@@ -1224,6 +1232,7 @@
}
if (request.haveResultMetadata && request.numBuffersLeft == 0) {
+ ATRACE_ASYNC_END("frame capture", frameNumber);
mInFlightMap.removeItemsAt(idx, 1);
}
@@ -1274,8 +1283,7 @@
if (entry.count == 0) {
SET_ERR("No timestamp provided by HAL for frame %d!",
frameNumber);
- }
- if (timestamp != entry.data.i64[0]) {
+ } else if (timestamp != entry.data.i64[0]) {
SET_ERR("Timestamp mismatch between shutter notify and result"
" metadata for frame %d (%lld vs %lld respectively)",
frameNumber, timestamp, entry.data.i64[0]);
@@ -1374,6 +1382,7 @@
}
void Camera3Device::notify(const camera3_notify_msg *msg) {
+ ATRACE_CALL();
NotificationListener *listener;
{
Mutex::Autolock l(mOutputLock);
@@ -1394,6 +1403,9 @@
msg->message.error.error_stream);
streamId = stream->getId();
}
+ ALOGV("Camera %d: %s: HAL error, frame %d, stream %d: %d",
+ mId, __FUNCTION__, msg->message.error.frame_number,
+ streamId, msg->message.error.error_code);
if (listener != NULL) {
listener->notifyError(msg->message.error.error_code,
msg->message.error.frame_number, streamId);
@@ -1429,7 +1441,8 @@
frameNumber);
break;
}
-
+ ALOGVV("Camera %d: %s: Shutter fired for frame %d at %lld",
+ mId, __FUNCTION__, frameNumber, timestamp);
// Call listener, if any
if (listener != NULL) {
listener->notifyShutter(frameNumber, timestamp);
@@ -1550,6 +1563,7 @@
}
status_t Camera3Device::RequestThread::waitUntilPaused(nsecs_t timeout) {
+ ATRACE_CALL();
status_t res;
Mutex::Autolock l(mPauseLock);
while (!mPaused) {
@@ -1696,8 +1710,11 @@
}
// Submit request and block until ready for next one
-
+ ATRACE_ASYNC_BEGIN("frame capture", request.frame_number);
+ ATRACE_BEGIN("camera3->process_capture_request");
res = mHal3Device->ops->process_capture_request(mHal3Device, &request);
+ ATRACE_END();
+
if (res != OK) {
SET_ERR("RequestThread: Unable to submit capture request %d to HAL"
" device: %s (%d)", request.frame_number, strerror(-res), res);
diff --git a/services/camera/libcameraservice/CameraClient.cpp b/services/camera/libcameraservice/CameraClient.cpp
index e577fa3..be78f69 100644
--- a/services/camera/libcameraservice/CameraClient.cpp
+++ b/services/camera/libcameraservice/CameraClient.cpp
@@ -347,6 +347,12 @@
}
}
+status_t CameraClient::setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer) {
+ ALOGE("%s: Unimplemented!", __FUNCTION__);
+ return INVALID_OPERATION;
+}
+
// start preview mode
status_t CameraClient::startPreview() {
LOG1("startPreview (pid %d)", getCallingPid());
diff --git a/services/camera/libcameraservice/CameraClient.h b/services/camera/libcameraservice/CameraClient.h
index 7f0cb29..abde75a 100644
--- a/services/camera/libcameraservice/CameraClient.h
+++ b/services/camera/libcameraservice/CameraClient.h
@@ -40,6 +40,8 @@
virtual status_t setPreviewDisplay(const sp<Surface>& surface);
virtual status_t setPreviewTexture(const sp<IGraphicBufferProducer>& bufferProducer);
virtual void setPreviewCallbackFlag(int flag);
+ virtual status_t setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer);
virtual status_t startPreview();
virtual void stopPreview();
virtual bool previewEnabled();
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 710f164..eaa316a 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -190,6 +190,8 @@
virtual status_t setPreviewDisplay(const sp<Surface>& surface) = 0;
virtual status_t setPreviewTexture(const sp<IGraphicBufferProducer>& bufferProducer)=0;
virtual void setPreviewCallbackFlag(int flag) = 0;
+ virtual status_t setPreviewCallbackTarget(
+ const sp<IGraphicBufferProducer>& callbackProducer) = 0;
virtual status_t startPreview() = 0;
virtual void stopPreview() = 0;
virtual bool previewEnabled() = 0;
diff --git a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
index 98673ff..bc81409 100644
--- a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
+++ b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
@@ -37,6 +37,7 @@
mDevice(client->getCameraDevice()),
mId(client->getCameraId()),
mCallbackAvailable(false),
+ mCallbackToApp(false),
mCallbackStreamId(NO_STREAM) {
}
@@ -53,6 +54,35 @@
}
}
+status_t CallbackProcessor::setCallbackWindow(
+ sp<ANativeWindow> callbackWindow) {
+ ATRACE_CALL();
+ status_t res;
+
+ Mutex::Autolock l(mInputMutex);
+
+ sp<Camera2Client> client = mClient.promote();
+ if (client == 0) return OK;
+ sp<CameraDeviceBase> device = client->getCameraDevice();
+
+ // If the window is changing, clear out stream if it already exists
+ if (mCallbackWindow != callbackWindow && mCallbackStreamId != NO_STREAM) {
+ res = device->deleteStream(mCallbackStreamId);
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to delete old stream "
+ "for callbacks: %s (%d)", __FUNCTION__,
+ client->getCameraId(), strerror(-res), res);
+ return res;
+ }
+ mCallbackStreamId = NO_STREAM;
+ mCallbackConsumer.clear();
+ }
+ mCallbackWindow = callbackWindow;
+ mCallbackToApp = (mCallbackWindow != NULL);
+
+ return OK;
+}
+
status_t CallbackProcessor::updateStream(const Parameters ¶ms) {
ATRACE_CALL();
status_t res;
@@ -67,14 +97,18 @@
// If possible, use the flexible YUV format
int32_t callbackFormat = params.previewFormat;
- if (params.fastInfo.useFlexibleYuv &&
+ if (mCallbackToApp) {
+ // TODO: etalvala: This should use the flexible YUV format as well, but
+ // need to reconcile HAL2/HAL3 requirements.
+ callbackFormat = HAL_PIXEL_FORMAT_YV12;
+ } else if(params.fastInfo.useFlexibleYuv &&
(params.previewFormat == HAL_PIXEL_FORMAT_YCrCb_420_SP ||
params.previewFormat == HAL_PIXEL_FORMAT_YV12) ) {
callbackFormat = HAL_PIXEL_FORMAT_YCbCr_420_888;
}
- if (mCallbackConsumer == 0) {
- // Create CPU buffer queue endpoint
+ if (!mCallbackToApp && mCallbackConsumer == 0) {
+ // Create CPU buffer queue endpoint, since app hasn't given us one
mCallbackConsumer = new CpuConsumer(kCallbackHeapCount);
mCallbackConsumer->setFrameAvailableListener(this);
mCallbackConsumer->setName(String8("Camera2Client::CallbackConsumer"));
@@ -104,8 +138,8 @@
res = device->deleteStream(mCallbackStreamId);
if (res != OK) {
ALOGE("%s: Camera %d: Unable to delete old output stream "
- "for callbacks: %s (%d)", __FUNCTION__, mId,
- strerror(-res), res);
+ "for callbacks: %s (%d)", __FUNCTION__,
+ mId, strerror(-res), res);
return res;
}
mCallbackStreamId = NO_STREAM;
diff --git a/services/camera/libcameraservice/camera2/CallbackProcessor.h b/services/camera/libcameraservice/camera2/CallbackProcessor.h
index d851a84..17dcfb1 100644
--- a/services/camera/libcameraservice/camera2/CallbackProcessor.h
+++ b/services/camera/libcameraservice/camera2/CallbackProcessor.h
@@ -45,6 +45,8 @@
void onFrameAvailable();
+ // Set to NULL to disable the direct-to-app callback window
+ status_t setCallbackWindow(sp<ANativeWindow> callbackWindow);
status_t updateStream(const Parameters ¶ms);
status_t deleteStream();
int getStreamId() const;
@@ -64,6 +66,9 @@
NO_STREAM = -1
};
+ // True if mCallbackWindow is a remote consumer, false if just the local
+ // mCallbackConsumer
+ bool mCallbackToApp;
int mCallbackStreamId;
static const size_t kCallbackHeapCount = 6;
sp<CpuConsumer> mCallbackConsumer;
diff --git a/services/camera/libcameraservice/camera2/JpegCompressor.cpp b/services/camera/libcameraservice/camera2/JpegCompressor.cpp
index c9af71e..2f0c67d 100644
--- a/services/camera/libcameraservice/camera2/JpegCompressor.cpp
+++ b/services/camera/libcameraservice/camera2/JpegCompressor.cpp
@@ -210,7 +210,8 @@
return true;
}
-void JpegCompressor::jpegTermDestination(j_compress_ptr /*cinfo*/) {
+void JpegCompressor::jpegTermDestination(j_compress_ptr cinfo) {
+ (void) cinfo; // TODO: clean up
ALOGV("%s", __FUNCTION__);
ALOGV("%s: Done writing JPEG data. %d bytes left in buffer",
__FUNCTION__, cinfo->dest->free_in_buffer);
diff --git a/services/camera/libcameraservice/camera2/Parameters.cpp b/services/camera/libcameraservice/camera2/Parameters.cpp
index a248b76..0459866 100644
--- a/services/camera/libcameraservice/camera2/Parameters.cpp
+++ b/services/camera/libcameraservice/camera2/Parameters.cpp
@@ -292,8 +292,11 @@
CameraParameters::WHITE_BALANCE_AUTO);
camera_metadata_ro_entry_t availableWhiteBalanceModes =
- staticInfo(ANDROID_CONTROL_AWB_AVAILABLE_MODES);
- {
+ staticInfo(ANDROID_CONTROL_AWB_AVAILABLE_MODES, 0, 0, false);
+ if (!availableWhiteBalanceModes.count) {
+ params.set(CameraParameters::KEY_SUPPORTED_WHITE_BALANCE,
+ CameraParameters::WHITE_BALANCE_AUTO);
+ } else {
String8 supportedWhiteBalance;
bool addComma = false;
for (size_t i=0; i < availableWhiteBalanceModes.count; i++) {
@@ -353,9 +356,11 @@
CameraParameters::EFFECT_NONE);
camera_metadata_ro_entry_t availableEffects =
- staticInfo(ANDROID_CONTROL_AVAILABLE_EFFECTS);
- if (!availableEffects.count) return NO_INIT;
- {
+ staticInfo(ANDROID_CONTROL_AVAILABLE_EFFECTS, 0, 0, false);
+ if (!availableEffects.count) {
+ params.set(CameraParameters::KEY_SUPPORTED_EFFECTS,
+ CameraParameters::EFFECT_NONE);
+ } else {
String8 supportedEffects;
bool addComma = false;
for (size_t i=0; i < availableEffects.count; i++) {
@@ -413,9 +418,11 @@
CameraParameters::ANTIBANDING_AUTO);
camera_metadata_ro_entry_t availableAntibandingModes =
- staticInfo(ANDROID_CONTROL_AE_AVAILABLE_ANTIBANDING_MODES);
- if (!availableAntibandingModes.count) return NO_INIT;
- {
+ staticInfo(ANDROID_CONTROL_AE_AVAILABLE_ANTIBANDING_MODES, 0, 0, false);
+ if (!availableAntibandingModes.count) {
+ params.set(CameraParameters::KEY_SUPPORTED_ANTIBANDING,
+ CameraParameters::ANTIBANDING_OFF);
+ } else {
String8 supportedAntibanding;
bool addComma = false;
for (size_t i=0; i < availableAntibandingModes.count; i++) {
@@ -455,9 +462,10 @@
CameraParameters::SCENE_MODE_AUTO);
camera_metadata_ro_entry_t availableSceneModes =
- staticInfo(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
- if (!availableSceneModes.count) return NO_INIT;
- {
+ staticInfo(ANDROID_CONTROL_AVAILABLE_SCENE_MODES, 0, 0, false);
+ if (!availableSceneModes.count) {
+ params.remove(CameraParameters::KEY_SCENE_MODE);
+ } else {
String8 supportedSceneModes(CameraParameters::SCENE_MODE_AUTO);
bool addComma = true;
bool noSceneModes = false;
@@ -548,15 +556,17 @@
}
}
+ bool isFlashAvailable = false;
camera_metadata_ro_entry_t flashAvailable =
- staticInfo(ANDROID_FLASH_INFO_AVAILABLE, 1, 1);
- if (!flashAvailable.count) return NO_INIT;
+ staticInfo(ANDROID_FLASH_INFO_AVAILABLE, 0, 1, false);
+ if (flashAvailable.count) {
+ isFlashAvailable = flashAvailable.data.u8[0];
+ }
camera_metadata_ro_entry_t availableAeModes =
- staticInfo(ANDROID_CONTROL_AE_AVAILABLE_MODES);
- if (!availableAeModes.count) return NO_INIT;
+ staticInfo(ANDROID_CONTROL_AE_AVAILABLE_MODES, 0, 0, false);
- if (flashAvailable.data.u8[0]) {
+ if (isFlashAvailable) {
flashMode = Parameters::FLASH_MODE_OFF;
params.set(CameraParameters::KEY_FLASH_MODE,
CameraParameters::FLASH_MODE_OFF);
@@ -585,14 +595,12 @@
}
camera_metadata_ro_entry_t minFocusDistance =
- staticInfo(ANDROID_LENS_INFO_MINIMUM_FOCUS_DISTANCE, 1, 1);
- if (!minFocusDistance.count) return NO_INIT;
+ staticInfo(ANDROID_LENS_INFO_MINIMUM_FOCUS_DISTANCE, 0, 1, false);
camera_metadata_ro_entry_t availableAfModes =
- staticInfo(ANDROID_CONTROL_AF_AVAILABLE_MODES);
- if (!availableAfModes.count) return NO_INIT;
+ staticInfo(ANDROID_CONTROL_AF_AVAILABLE_MODES, 0, 0, false);
- if (minFocusDistance.data.f[0] == 0) {
+ if (!minFocusDistance.count || minFocusDistance.data.f[0] == 0) {
// Fixed-focus lens
focusMode = Parameters::FOCUS_MODE_FIXED;
params.set(CameraParameters::KEY_FOCUS_MODE,
@@ -662,7 +670,7 @@
focusingAreas.add(Parameters::Area(0,0,0,0,0));
camera_metadata_ro_entry_t availableFocalLengths =
- staticInfo(ANDROID_LENS_INFO_AVAILABLE_FOCAL_LENGTHS);
+ staticInfo(ANDROID_LENS_INFO_AVAILABLE_FOCAL_LENGTHS, 0, 0, false);
if (!availableFocalLengths.count) return NO_INIT;
float minFocalLength = availableFocalLengths.data.f[0];
@@ -768,8 +776,8 @@
CameraParameters::FALSE);
camera_metadata_ro_entry_t availableVideoStabilizationModes =
- staticInfo(ANDROID_CONTROL_AVAILABLE_VIDEO_STABILIZATION_MODES);
- if (!availableVideoStabilizationModes.count) return NO_INIT;
+ staticInfo(ANDROID_CONTROL_AVAILABLE_VIDEO_STABILIZATION_MODES, 0, 0,
+ false);
if (availableVideoStabilizationModes.count > 1) {
params.set(CameraParameters::KEY_VIDEO_STABILIZATION_SUPPORTED,
@@ -794,9 +802,10 @@
previewCallbackFlags = 0;
previewCallbackOneShot = false;
+ previewCallbackSurface = false;
camera_metadata_ro_entry_t supportedHardwareLevel =
- staticInfo(ANDROID_INFO_SUPPORTED_HARDWARE_LEVEL);
+ staticInfo(ANDROID_INFO_SUPPORTED_HARDWARE_LEVEL, 0, 0, false);
if (!supportedHardwareLevel.count || (supportedHardwareLevel.data.u8[0] ==
ANDROID_INFO_SUPPORTED_HARDWARE_LEVEL_LIMITED)) {
ALOGI("Camera %d: ZSL mode disabled for limited mode HALs", cameraId);
@@ -828,14 +837,23 @@
status_t Parameters::buildFastInfo() {
camera_metadata_ro_entry_t activeArraySize =
- staticInfo(ANDROID_SENSOR_INFO_ACTIVE_ARRAY_SIZE, 2, 2);
+ staticInfo(ANDROID_SENSOR_INFO_ACTIVE_ARRAY_SIZE, 2, 4);
if (!activeArraySize.count) return NO_INIT;
- int32_t arrayWidth = activeArraySize.data.i32[0];
- int32_t arrayHeight = activeArraySize.data.i32[1];
+ int32_t arrayWidth;
+ int32_t arrayHeight;
+ if (activeArraySize.count == 2) {
+ ALOGW("%s: Camera %d: activeArraySize is missing xmin/ymin!",
+ __FUNCTION__, cameraId);
+ arrayWidth = activeArraySize.data.i32[0];
+ arrayHeight = activeArraySize.data.i32[1];
+ } else if (activeArraySize.count == 4) {
+ arrayWidth = activeArraySize.data.i32[2];
+ arrayHeight = activeArraySize.data.i32[3];
+ } else return NO_INIT;
camera_metadata_ro_entry_t availableFaceDetectModes =
- staticInfo(ANDROID_STATISTICS_INFO_AVAILABLE_FACE_DETECT_MODES);
- if (!availableFaceDetectModes.count) return NO_INIT;
+ staticInfo(ANDROID_STATISTICS_INFO_AVAILABLE_FACE_DETECT_MODES, 0, 0,
+ false);
uint8_t bestFaceDetectMode =
ANDROID_STATISTICS_FACE_DETECT_MODE_OFF;
@@ -862,19 +880,21 @@
}
}
+ int32_t maxFaces = 0;
camera_metadata_ro_entry_t maxFacesDetected =
- staticInfo(ANDROID_STATISTICS_INFO_MAX_FACE_COUNT, 1, 1);
- if (!maxFacesDetected.count) return NO_INIT;
-
- int32_t maxFaces = maxFacesDetected.data.i32[0];
+ staticInfo(ANDROID_STATISTICS_INFO_MAX_FACE_COUNT, 0, 1, false);
+ if (maxFacesDetected.count) {
+ maxFaces = maxFacesDetected.data.i32[0];
+ }
camera_metadata_ro_entry_t availableSceneModes =
- staticInfo(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
+ staticInfo(ANDROID_CONTROL_AVAILABLE_SCENE_MODES, 0, 0, false);
camera_metadata_ro_entry_t sceneModeOverrides =
- staticInfo(ANDROID_CONTROL_SCENE_MODE_OVERRIDES);
+ staticInfo(ANDROID_CONTROL_SCENE_MODE_OVERRIDES, 0, 0, false);
camera_metadata_ro_entry_t minFocusDistance =
- staticInfo(ANDROID_LENS_INFO_MINIMUM_FOCUS_DISTANCE);
- bool fixedLens = (minFocusDistance.data.f[0] == 0);
+ staticInfo(ANDROID_LENS_INFO_MINIMUM_FOCUS_DISTANCE, 0, 0, false);
+ bool fixedLens = minFocusDistance.count == 0 ||
+ minFocusDistance.data.f[0] == 0;
camera_metadata_ro_entry_t availableFocalLengths =
staticInfo(ANDROID_LENS_INFO_AVAILABLE_FOCAL_LENGTHS);
@@ -1465,7 +1485,7 @@
}
if (validatedParams.wbMode != wbMode) {
camera_metadata_ro_entry_t availableWbModes =
- staticInfo(ANDROID_CONTROL_AWB_AVAILABLE_MODES);
+ staticInfo(ANDROID_CONTROL_AWB_AVAILABLE_MODES, 0, 0, false);
for (i = 0; i < availableWbModes.count; i++) {
if (validatedParams.wbMode == availableWbModes.data.u8[i]) break;
}
@@ -1496,8 +1516,9 @@
validatedParams.currentAfTriggerId = -1;
if (validatedParams.focusMode != Parameters::FOCUS_MODE_FIXED) {
camera_metadata_ro_entry_t minFocusDistance =
- staticInfo(ANDROID_LENS_INFO_MINIMUM_FOCUS_DISTANCE);
- if (minFocusDistance.data.f[0] == 0) {
+ staticInfo(ANDROID_LENS_INFO_MINIMUM_FOCUS_DISTANCE, 0, 0,
+ false);
+ if (minFocusDistance.count && minFocusDistance.data.f[0] == 0) {
ALOGE("%s: Requested focus mode \"%s\" is not available: "
"fixed focus lens",
__FUNCTION__,
@@ -1617,7 +1638,8 @@
validatedParams.videoStabilization = boolFromString(
newParams.get(CameraParameters::KEY_VIDEO_STABILIZATION) );
camera_metadata_ro_entry_t availableVideoStabilizationModes =
- staticInfo(ANDROID_CONTROL_AVAILABLE_VIDEO_STABILIZATION_MODES);
+ staticInfo(ANDROID_CONTROL_AVAILABLE_VIDEO_STABILIZATION_MODES, 0, 0,
+ false);
if (validatedParams.videoStabilization &&
availableVideoStabilizationModes.count == 1) {
ALOGE("%s: Video stabilization not supported", __FUNCTION__);
@@ -2544,10 +2566,6 @@
staticInfo(ANDROID_SENSOR_INFO_PHYSICAL_SIZE, 2, 2);
if (!sensorSize.count) return NO_INIT;
- camera_metadata_ro_entry_t availableFocalLengths =
- staticInfo(ANDROID_LENS_INFO_AVAILABLE_FOCAL_LENGTHS);
- if (!availableFocalLengths.count) return NO_INIT;
-
float arrayAspect = static_cast<float>(fastInfo.arrayWidth) /
fastInfo.arrayHeight;
float stillAspect = static_cast<float>(pictureWidth) / pictureHeight;
diff --git a/services/camera/libcameraservice/camera2/Parameters.h b/services/camera/libcameraservice/camera2/Parameters.h
index be05b54..464830c 100644
--- a/services/camera/libcameraservice/camera2/Parameters.h
+++ b/services/camera/libcameraservice/camera2/Parameters.h
@@ -142,6 +142,7 @@
uint32_t previewCallbackFlags;
bool previewCallbackOneShot;
+ bool previewCallbackSurface;
bool zslMode;
diff --git a/services/camera/libcameraservice/camera3/Camera3OutputStream.cpp b/services/camera/libcameraservice/camera3/Camera3OutputStream.cpp
index 2efeede..f085443 100644
--- a/services/camera/libcameraservice/camera3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/camera3/Camera3OutputStream.cpp
@@ -301,8 +301,13 @@
return res;
}
- ALOGV("%s: Consumer wants %d buffers", __FUNCTION__,
- maxConsumerBuffers);
+ ALOGV("%s: Consumer wants %d buffers, HAL wants %d", __FUNCTION__,
+ maxConsumerBuffers, camera3_stream::max_buffers);
+ if (camera3_stream::max_buffers == 0) {
+ ALOGE("%s: Camera HAL requested no max_buffers, requires at least 1",
+ __FUNCTION__, camera3_stream::max_buffers);
+ return INVALID_OPERATION;
+ }
mTotalBufferCount = maxConsumerBuffers + camera3_stream::max_buffers;
mDequeuedBufferCount = 0;
diff --git a/services/camera/libcameraservice/camera3/Camera3Stream.cpp b/services/camera/libcameraservice/camera3/Camera3Stream.cpp
index f05658a..ab563df 100644
--- a/services/camera/libcameraservice/camera3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/camera3/Camera3Stream.cpp
@@ -312,8 +312,10 @@
// Got all buffers, register with HAL
ALOGV("%s: Registering %d buffers with camera HAL",
__FUNCTION__, bufferCount);
+ ATRACE_BEGIN("camera3->register_stream_buffers");
res = hal3Device->ops->register_stream_buffers(hal3Device,
&bufferSet);
+ ATRACE_END();
}
// Return all valid buffers to stream, in ERROR state to indicate
diff --git a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
index cd39bad..dfa1066 100644
--- a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
+++ b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
@@ -214,7 +214,11 @@
// In case the object was never pinned, pass the acquire fence
// back to the release fence. If the fence was already waited on,
// it'll just be a no-op to wait on it again.
- err = addReleaseFenceLocked(item.mBuf, item.mFence);
+
+ // item.mGraphicBuffer was populated with the proper graphic-buffer
+ // at acquire even if it was previously acquired
+ err = addReleaseFenceLocked(item.mBuf,
+ item.mGraphicBuffer, item.mFence);
if (err != OK) {
BI_LOGE("Failed to add release fence to buffer "
@@ -226,7 +230,9 @@
BI_LOGV("Attempting to release buffer timestamp %lld, frame %lld",
item.mTimestamp, item.mFrameNumber);
- err = releaseBufferLocked(item.mBuf,
+ // item.mGraphicBuffer was populated with the proper graphic-buffer
+ // at acquire even if it was previously acquired
+ err = releaseBufferLocked(item.mBuf, item.mGraphicBuffer,
EGL_NO_DISPLAY,
EGL_NO_SYNC_KHR);
if (err != OK) {
@@ -310,7 +316,8 @@
RingBufferItem& find = *it;
if (item.mGraphicBuffer == find.mGraphicBuffer) {
- status_t res = addReleaseFenceLocked(item.mBuf, item.mFence);
+ status_t res = addReleaseFenceLocked(item.mBuf,
+ item.mGraphicBuffer, item.mFence);
if (res != OK) {
BI_LOGE("Failed to add release fence to buffer "