Merge "AudioRecord::getInputFramesLost() cleanup"
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 80cef8d..c724949 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -39,8 +39,12 @@
* Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
*/
enum event_type {
- EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer.
- EVENT_OVERRUN = 1, // PCM buffer overrun occurred.
+ EVENT_MORE_DATA = 0, // Request to read available data from buffer.
+ // If this event is delivered but the callback handler
+ // does not want to read the available data, the handler must
+ // explicitly
+ // ignore the event by setting frameCount to zero.
+ EVENT_OVERRUN = 1, // Buffer overrun occurred.
EVENT_MARKER = 2, // Record head is at the specified marker position
// (See setMarkerPosition()).
EVENT_NEW_POS = 3, // Record head is at a new position
@@ -63,6 +67,7 @@
// (currently ignored but will make the primary field in future)
size_t size; // input/output in bytes == frameCount * frameSize
+ // on output is the number of bytes actually drained
// 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
@@ -76,7 +81,7 @@
/* 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 ready or various conditions occur.
+ * invokes the callback when a new buffer becomes available or various conditions occur.
* Parameters:
*
* event: type of event notified (see enum AudioRecord::event_type).
@@ -99,6 +104,8 @@
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
* - BAD_VALUE: unsupported configuration
+ * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+ * and is undefined otherwise.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -109,7 +116,7 @@
/* How data is transferred from AudioRecord
*/
enum transfer_type {
- TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters
+ 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 read()
@@ -137,7 +144,7 @@
* 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 and inform of marker, position updates, etc.
+ * to consume new 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.
@@ -171,9 +178,10 @@
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful intialization
* - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
- * - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+ * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
+ * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
*
* Parameters not listed in the AudioRecord constructors above:
*
@@ -192,7 +200,7 @@
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
- /* Result of constructing the AudioRecord. This must be checked
+ /* Result of constructing the AudioRecord. This must be checked for successful initialization
* before using any AudioRecord API (except for set()), because using
* an uninitialized AudioRecord produces undefined results.
* See set() method above for possible return codes.
@@ -221,7 +229,7 @@
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. Note that obtainBuffer() still
+ /* Stop a track. 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();
@@ -236,7 +244,7 @@
* 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.
+ * a workaround is to 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.
*
@@ -428,6 +436,7 @@
nsecs_t processAudioBuffer();
// caller must hold lock on mLock for all _l methods
+
status_t openRecord_l(size_t epoch);
// FIXME enum is faster than strcmp() for parameter 'from'
@@ -479,7 +488,7 @@
audio_io_handle_t mInput; // returned by AudioSystem::getInput()
- // may be changed if IAudioRecord object is re-created
+ // Next 3 fields may be changed if IAudioRecord is re-created, but always != 0
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 96135a6..88a5a4c 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -160,7 +160,7 @@
* 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.
+ * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
* frameCount: Minimum size of track PCM buffer in frames. This defines the
* application's contribution to the
* latency of the track. The actual size selected by the AudioTrack could be
@@ -338,7 +338,7 @@
*/
status_t setSampleRate(uint32_t sampleRate);
- /* Return current source sample rate in Hz, or 0 if unknown */
+ /* Return current source sample rate in Hz */
uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
@@ -363,7 +363,7 @@
/* 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 ~0 or 1.
+ * a workaround is to 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.
*
@@ -664,9 +664,10 @@
float mVolume[2];
float mSendLevel;
mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it.
- size_t mFrameCount; // corresponds to current IAudioTrack
- size_t mReqFrameCount; // frame count to request the next time a new
- // IAudioTrack is needed
+ size_t mFrameCount; // corresponds to current IAudioTrack, value is
+ // reported back by AudioFlinger to the client
+ size_t mReqFrameCount; // frame count to request the first or next time
+ // a new IAudioTrack is needed, non-decreasing
// constant after constructor or set()
audio_format_t mFormat; // as requested by client, not forced to 16-bit
@@ -716,6 +717,7 @@
sp<IMemory> mSharedBuffer;
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
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 0a728a8..da513c8 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -352,6 +352,7 @@
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
+ // The only purpose of setting marker position is to get a callback
if (mCbf == NULL) {
return INVALID_OPERATION;
}
@@ -377,6 +378,7 @@
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
+ // The only purpose of setting position update period is to get a callback
if (mCbf == NULL) {
return INVALID_OPERATION;
}
@@ -770,7 +772,7 @@
int32_t sequence = mSequence;
// These fields don't need to be cached, because they are assigned only by set():
- // mTransfer, mCbf, mUserData, mSampleRate
+ // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
mLock.unlock();
@@ -844,8 +846,8 @@
"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);
+ ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+ mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
break;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 2188cac..dcb72f8 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -735,7 +735,8 @@
audio_io_handle_t AudioSystem::getOutputForEffect(const effect_descriptor_t *desc)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return PERMISSION_DENIED;
+ // FIXME change return type to status_t, and return PERMISSION_DENIED here
+ if (aps == 0) return 0;
return aps->getOutputForEffect(desc);
}
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 8954d9f..f5a010b 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -320,7 +320,7 @@
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0.0f;
- mFrameCount = frameCount;
+ // mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
@@ -355,7 +355,7 @@
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
- //Use of direct and offloaded output streams is ref counted by audio policy manager.
+ // Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
AudioSystem::releaseOutput(output);
@@ -698,6 +698,7 @@
AutoMutex lock(mLock);
mNewPosition = mProxy->getPosition() + updatePeriod;
mUpdatePeriod = updatePeriod;
+
return NO_ERROR;
}
@@ -1735,7 +1736,7 @@
}
}
if (result != NO_ERROR) {
- //Use of direct and offloaded output streams is ref counted by audio policy manager.
+ // Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
AudioSystem::releaseOutput(output);
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 130207d..aae6800 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -2096,7 +2096,10 @@
mSeekNotificationSent = true;
}
- mSeeking = NO_SEEK;
+ if (mVideoSource == NULL) {
+ // For video the mSeeking flag is always reset in finishSeekIfNecessary
+ mSeeking = NO_SEEK;
+ }
notifyIfMediaStarted_l();
}
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index c8f0730..415f696 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -475,8 +475,9 @@
audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc)
{
+ // FIXME change return type to status_t, and return NO_INIT here
if (mpAudioPolicy == NULL) {
- return NO_INIT;
+ return 0;
}
Mutex::Autolock _l(mLock);
return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc);
diff --git a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
index ebc7ea7..9a6dc28 100644
--- a/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
+++ b/services/camera/libcameraservice/gui/RingBufferConsumer.cpp
@@ -21,11 +21,11 @@
#include <gui/RingBufferConsumer.h>
-#define BI_LOGV(x, ...) ALOGV("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGD(x, ...) ALOGD("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGI(x, ...) ALOGI("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGW(x, ...) ALOGW("[%s] "x, mName.string(), ##__VA_ARGS__)
-#define BI_LOGE(x, ...) ALOGE("[%s] "x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGV(x, ...) ALOGV("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGD(x, ...) ALOGD("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGI(x, ...) ALOGI("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGW(x, ...) ALOGW("[%s] " x, mName.string(), ##__VA_ARGS__)
+#define BI_LOGE(x, ...) ALOGE("[%s] " x, mName.string(), ##__VA_ARGS__)
#undef assert
#define assert(x) ALOG_ASSERT((x), #x)