Merge "libstagefright: Do not use max/min luminance if it's 0" into rvc-qpr-dev
diff --git a/media/codec2/components/gav1/C2SoftGav1Dec.cpp b/media/codec2/components/gav1/C2SoftGav1Dec.cpp
index cf7c370..120ba7a 100644
--- a/media/codec2/components/gav1/C2SoftGav1Dec.cpp
+++ b/media/codec2/components/gav1/C2SoftGav1Dec.cpp
@@ -446,15 +446,6 @@
return;
}
- } else {
- const Libgav1StatusCode status = mCodecCtx->SignalEOS();
- if (status != kLibgav1StatusOk) {
- ALOGE("Failed to flush av1 decoder. status: %d.", status);
- work->result = C2_CORRUPTED;
- work->workletsProcessed = 1u;
- mSignalledError = true;
- return;
- }
}
(void)outputBuffer(pool, work);
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index a47f189..cab8929 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -1035,6 +1035,11 @@
* but still allow queries to the stream to occur from other threads. This often
* happens if you are monitoring stream progress from a UI thread.
*
+ * NOTE: This function is only fully implemented for MMAP streams,
+ * which are low latency streams supported by some devices.
+ * On other "Legacy" streams some audio resources will still be in use
+ * and some callbacks may still be in process after this call.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 9fa2e40..fce322b 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -14,8 +14,6 @@
* limitations under the License.
*/
-#define LOG_TAG (mInService ? "AudioStreamInternalCapture_Service" \
- : "AudioStreamInternalCapture_Client")
//#define LOG_NDEBUG 0
#include <utils/Log.h>
@@ -29,6 +27,14 @@
#define ATRACE_TAG ATRACE_TAG_AUDIO
#include <utils/Trace.h>
+// We do this after the #includes because if a header uses ALOG.
+// it would fail on the reference to mInService.
+#undef LOG_TAG
+// This file is used in both client and server processes.
+// This is needed to make sense of the logs more easily.
+#define LOG_TAG (mInService ? "AudioStreamInternalCapture_Service" \
+ : "AudioStreamInternalCapture_Client")
+
using android::WrappingBuffer;
using namespace aaudio;
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 1303daf..d6b73b4 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -14,8 +14,6 @@
* limitations under the License.
*/
-#define LOG_TAG (mInService ? "AudioStreamInternalPlay_Service" \
- : "AudioStreamInternalPlay_Client")
//#define LOG_NDEBUG 0
#include <utils/Log.h>
@@ -26,6 +24,14 @@
#include "client/AudioStreamInternalPlay.h"
#include "utility/AudioClock.h"
+// We do this after the #includes because if a header uses ALOG.
+// it would fail on the reference to mInService.
+#undef LOG_TAG
+// This file is used in both client and server processes.
+// This is needed to make sense of the logs more easily.
+#define LOG_TAG (mInService ? "AudioStreamInternalPlay_Service" \
+ : "AudioStreamInternalPlay_Client")
+
using android::WrappingBuffer;
using namespace aaudio;
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 8965875..cfa7221 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -255,17 +255,16 @@
if (audioStream != nullptr) {
aaudio_stream_id_t id = audioStream->getId();
ALOGD("%s(s#%u) called ---------------", __func__, id);
- result = audioStream->safeRelease();
- // safeRelease will only fail if called illegally, for example, from a callback.
+ result = audioStream->safeReleaseClose();
+ // safeReleaseClose will only fail if called illegally, for example, from a callback.
// That would result in deleting an active stream, which would cause a crash.
if (result != AAUDIO_OK) {
ALOGW("%s(s#%u) failed. Close it from another thread.",
__func__, id);
} else {
audioStream->unregisterPlayerBase();
- // Mark CLOSED to keep destructors from asserting.
- audioStream->closeFinal();
- delete audioStream;
+ // Allow the stream to be deleted.
+ AudioStreamBuilder::stopUsingStream(audioStream);
}
ALOGD("%s(s#%u) returned %d ---------", __func__, id, result);
}
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index f5c75ca..43240ec 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -39,7 +39,7 @@
}
AudioStream::AudioStream()
- : mPlayerBase(new MyPlayerBase(this))
+ : mPlayerBase(new MyPlayerBase())
, mStreamId(AAudio_getNextStreamId())
{
// mThread is a pthread_t of unknown size so we need memset.
@@ -48,6 +48,10 @@
}
AudioStream::~AudioStream() {
+ // Please preserve this log because there have been several bugs related to
+ // AudioStream deletion and late callbacks.
+ ALOGD("%s(s#%u) mPlayerBase strongCount = %d",
+ __func__, getId(), mPlayerBase->getStrongCount());
// If the stream is deleted when OPEN or in use then audio resources will leak.
// This would indicate an internal error. So we want to find this ASAP.
LOG_ALWAYS_FATAL_IF(!(getState() == AAUDIO_STREAM_STATE_CLOSED
@@ -55,8 +59,6 @@
|| getState() == AAUDIO_STREAM_STATE_DISCONNECTED),
"~AudioStream() - still in use, state = %s",
AudioGlobal_convertStreamStateToText(getState()));
-
- mPlayerBase->clearParentReference(); // remove reference to this AudioStream
}
aaudio_result_t AudioStream::open(const AudioStreamBuilder& builder)
@@ -301,18 +303,29 @@
}
aaudio_result_t AudioStream::safeRelease() {
- // This get temporarily unlocked in the release() when joining callback threads.
+ // This get temporarily unlocked in the MMAP release() when joining callback threads.
std::lock_guard<std::mutex> lock(mStreamLock);
if (collidesWithCallback()) {
ALOGE("%s cannot be called from a callback!", __func__);
return AAUDIO_ERROR_INVALID_STATE;
}
- if (getState() == AAUDIO_STREAM_STATE_CLOSING) {
+ if (getState() == AAUDIO_STREAM_STATE_CLOSING) { // already released?
return AAUDIO_OK;
}
return release_l();
}
+aaudio_result_t AudioStream::safeReleaseClose() {
+ // This get temporarily unlocked in the MMAP release() when joining callback threads.
+ std::lock_guard<std::mutex> lock(mStreamLock);
+ if (collidesWithCallback()) {
+ ALOGE("%s cannot be called from a callback!", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ releaseCloseFinal();
+ return AAUDIO_OK;
+}
+
void AudioStream::setState(aaudio_stream_state_t state) {
ALOGD("%s(s#%d) from %d to %d", __func__, getId(), mState, state);
// Track transition to DISCONNECTED state.
@@ -520,11 +533,18 @@
}
#if AAUDIO_USE_VOLUME_SHAPER
-android::media::VolumeShaper::Status AudioStream::applyVolumeShaper(
- const android::media::VolumeShaper::Configuration& configuration __unused,
- const android::media::VolumeShaper::Operation& operation __unused) {
- ALOGW("applyVolumeShaper() is not supported");
- return android::media::VolumeShaper::Status::ok();
+::android::binder::Status AudioStream::MyPlayerBase::applyVolumeShaper(
+ const ::android::media::VolumeShaper::Configuration& configuration,
+ const ::android::media::VolumeShaper::Operation& operation) {
+ android::sp<AudioStream> audioStream;
+ {
+ std::lock_guard<std::mutex> lock(mParentLock);
+ audioStream = mParent.promote();
+ }
+ if (audioStream) {
+ return audioStream->applyVolumeShaper(configuration, operation);
+ }
+ return android::NO_ERROR;
}
#endif
@@ -534,26 +554,36 @@
doSetVolume(); // apply this change
}
-AudioStream::MyPlayerBase::MyPlayerBase(AudioStream *parent) : mParent(parent) {
-}
-
-AudioStream::MyPlayerBase::~MyPlayerBase() {
-}
-
-void AudioStream::MyPlayerBase::registerWithAudioManager() {
+void AudioStream::MyPlayerBase::registerWithAudioManager(const android::sp<AudioStream>& parent) {
+ std::lock_guard<std::mutex> lock(mParentLock);
+ mParent = parent;
if (!mRegistered) {
- init(android::PLAYER_TYPE_AAUDIO, AAudioConvert_usageToInternal(mParent->getUsage()));
+ init(android::PLAYER_TYPE_AAUDIO, AAudioConvert_usageToInternal(parent->getUsage()));
mRegistered = true;
}
}
void AudioStream::MyPlayerBase::unregisterWithAudioManager() {
+ std::lock_guard<std::mutex> lock(mParentLock);
if (mRegistered) {
baseDestroy();
mRegistered = false;
}
}
+android::status_t AudioStream::MyPlayerBase::playerSetVolume() {
+ android::sp<AudioStream> audioStream;
+ {
+ std::lock_guard<std::mutex> lock(mParentLock);
+ audioStream = mParent.promote();
+ }
+ if (audioStream) {
+ // No pan and only left volume is taken into account from IPLayer interface
+ audioStream->setDuckAndMuteVolume(mVolumeMultiplierL /* * mPanMultiplierL */);
+ }
+ return android::NO_ERROR;
+}
+
void AudioStream::MyPlayerBase::destroy() {
unregisterWithAudioManager();
}
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index fb71c36..e0bd9d8 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -25,8 +25,10 @@
#include <binder/Status.h>
#include <utils/StrongPointer.h>
-#include "media/VolumeShaper.h"
-#include "media/PlayerBase.h"
+#include <media/AudioSystem.h>
+#include <media/PlayerBase.h>
+#include <media/VolumeShaper.h>
+
#include "utility/AAudioUtilities.h"
#include "utility/MonotonicCounter.h"
@@ -45,7 +47,8 @@
/**
* AAudio audio stream.
*/
-class AudioStream {
+// By extending AudioDeviceCallback, we also inherit from RefBase.
+class AudioStream : public android::AudioSystem::AudioDeviceCallback {
public:
AudioStream();
@@ -117,6 +120,17 @@
virtual void logOpen();
void logReleaseBufferState();
+ /* Note about naming for "release" and "close" related methods.
+ *
+ * These names are intended to match the public AAudio API.
+ * The original AAudio API had an AAudioStream_close() function that
+ * released the hardware and deleted the stream. That made it difficult
+ * because apps want to release the HW ASAP but are not in a rush to delete
+ * the stream object. So in R we added an AAudioStream_release() function
+ * that just released the hardware.
+ * The AAudioStream_close() method releases if needed and then closes.
+ */
+
/**
* Free any hardware or system resources from the open() call.
* It is safe to call release_l() multiple times.
@@ -126,22 +140,27 @@
return AAUDIO_OK;
}
- aaudio_result_t closeFinal() {
+ /**
+ * Free any resources not already freed by release_l().
+ * Assume release_l() already called.
+ */
+ virtual void close_l() {
+ // Releasing the stream will set the state to CLOSING.
+ assert(getState() == AAUDIO_STREAM_STATE_CLOSING);
+ // setState() prevents a transition from CLOSING to any state other than CLOSED.
// State is checked by destructor.
setState(AAUDIO_STREAM_STATE_CLOSED);
- return AAUDIO_OK;
}
/**
* Release then close the stream.
- * @return AAUDIO_OK or negative error.
*/
- aaudio_result_t releaseCloseFinal() {
- aaudio_result_t result = release_l(); // TODO review locking
- if (result == AAUDIO_OK) {
- result = closeFinal();
+ void releaseCloseFinal() {
+ if (getState() != AAUDIO_STREAM_STATE_CLOSING) { // not already released?
+ // Ignore result and keep closing.
+ (void) release_l();
}
- return result;
+ close_l();
}
// This is only used to identify a stream in the logs without
@@ -328,6 +347,10 @@
*/
bool collidesWithCallback() const;
+ // Implement AudioDeviceCallback
+ void onAudioDeviceUpdate(audio_io_handle_t audioIo,
+ audio_port_handle_t deviceId) override {};
+
// ============== I/O ===========================
// A Stream will only implement read() or write() depending on its direction.
virtual aaudio_result_t write(const void *buffer __unused,
@@ -366,7 +389,7 @@
*/
void registerPlayerBase() {
if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
- mPlayerBase->registerWithAudioManager();
+ mPlayerBase->registerWithAudioManager(this);
}
}
@@ -395,21 +418,33 @@
*/
aaudio_result_t systemStopFromCallback();
+ /**
+ * Safely RELEASE a stream after taking mStreamLock and checking
+ * to make sure we are not being called from a callback.
+ * @return AAUDIO_OK or a negative error
+ */
aaudio_result_t safeRelease();
+ /**
+ * Safely RELEASE and CLOSE a stream after taking mStreamLock and checking
+ * to make sure we are not being called from a callback.
+ * @return AAUDIO_OK or a negative error
+ */
+ aaudio_result_t safeReleaseClose();
+
protected:
// PlayerBase allows the system to control the stream volume.
class MyPlayerBase : public android::PlayerBase {
public:
- explicit MyPlayerBase(AudioStream *parent);
+ MyPlayerBase() {};
- virtual ~MyPlayerBase();
+ virtual ~MyPlayerBase() = default;
/**
* Register for volume changes and remote control.
*/
- void registerWithAudioManager();
+ void registerWithAudioManager(const android::sp<AudioStream>& parent);
/**
* UnRegister.
@@ -421,8 +456,6 @@
*/
void destroy() override;
- void clearParentReference() { mParent = nullptr; }
-
// Just a stub. The ability to start audio through PlayerBase is being deprecated.
android::status_t playerStart() override {
return android::NO_ERROR;
@@ -438,18 +471,10 @@
return android::NO_ERROR;
}
- android::status_t playerSetVolume() override {
- // No pan and only left volume is taken into account from IPLayer interface
- mParent->setDuckAndMuteVolume(mVolumeMultiplierL /* * mPanMultiplierL */);
- return android::NO_ERROR;
- }
+ android::status_t playerSetVolume() override;
#if AAUDIO_USE_VOLUME_SHAPER
- ::android::binder::Status applyVolumeShaper(
- const ::android::media::VolumeShaper::Configuration& configuration,
- const ::android::media::VolumeShaper::Operation& operation) {
- return mParent->applyVolumeShaper(configuration, operation);
- }
+ ::android::binder::Status applyVolumeShaper();
#endif
aaudio_result_t getResult() {
@@ -457,9 +482,12 @@
}
private:
- AudioStream *mParent;
- aaudio_result_t mResult = AAUDIO_OK;
- bool mRegistered = false;
+ // Use a weak pointer so the AudioStream can be deleted.
+
+ std::mutex mParentLock;
+ android::wp<AudioStream> mParent;
+ aaudio_result_t mResult = AAUDIO_OK;
+ bool mRegistered = false;
};
/**
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 60dad84..630b289 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -63,27 +63,26 @@
static aaudio_result_t builder_createStream(aaudio_direction_t direction,
aaudio_sharing_mode_t sharingMode,
bool tryMMap,
- AudioStream **audioStreamPtr) {
- *audioStreamPtr = nullptr;
+ android::sp<AudioStream> &stream) {
aaudio_result_t result = AAUDIO_OK;
switch (direction) {
case AAUDIO_DIRECTION_INPUT:
if (tryMMap) {
- *audioStreamPtr = new AudioStreamInternalCapture(AAudioBinderClient::getInstance(),
+ stream = new AudioStreamInternalCapture(AAudioBinderClient::getInstance(),
false);
} else {
- *audioStreamPtr = new AudioStreamRecord();
+ stream = new AudioStreamRecord();
}
break;
case AAUDIO_DIRECTION_OUTPUT:
if (tryMMap) {
- *audioStreamPtr = new AudioStreamInternalPlay(AAudioBinderClient::getInstance(),
+ stream = new AudioStreamInternalPlay(AAudioBinderClient::getInstance(),
false);
} else {
- *audioStreamPtr = new AudioStreamTrack();
+ stream = new AudioStreamTrack();
}
break;
@@ -98,7 +97,7 @@
// Fall back to Legacy path if MMAP not available.
// Exact behavior is controlled by MMapPolicy.
aaudio_result_t AudioStreamBuilder::build(AudioStream** streamPtr) {
- AudioStream *audioStream = nullptr;
+
if (streamPtr == nullptr) {
ALOGE("%s() streamPtr is null", __func__);
return AAUDIO_ERROR_NULL;
@@ -171,41 +170,48 @@
setPrivacySensitive(true);
}
- result = builder_createStream(getDirection(), sharingMode, allowMMap, &audioStream);
+ android::sp<AudioStream> audioStream;
+ result = builder_createStream(getDirection(), sharingMode, allowMMap, audioStream);
if (result == AAUDIO_OK) {
// Open the stream using the parameters from the builder.
result = audioStream->open(*this);
- if (result == AAUDIO_OK) {
- *streamPtr = audioStream;
- } else {
+ if (result != AAUDIO_OK) {
bool isMMap = audioStream->isMMap();
- delete audioStream;
- audioStream = nullptr;
-
if (isMMap && allowLegacy) {
ALOGV("%s() MMAP stream did not open so try Legacy path", __func__);
// If MMAP stream failed to open then TRY using a legacy stream.
result = builder_createStream(getDirection(), sharingMode,
- false, &audioStream);
+ false, audioStream);
if (result == AAUDIO_OK) {
result = audioStream->open(*this);
- if (result == AAUDIO_OK) {
- *streamPtr = audioStream;
- } else {
- delete audioStream;
- audioStream = nullptr;
- }
}
}
}
- if (audioStream != nullptr) {
+ if (result == AAUDIO_OK) {
audioStream->logOpen();
- }
+ *streamPtr = startUsingStream(audioStream);
+ } // else audioStream will go out of scope and be deleted
}
return result;
}
+AudioStream *AudioStreamBuilder::startUsingStream(android::sp<AudioStream> &audioStream) {
+ // Increment the smart pointer so it will not get deleted when
+ // we pass it to the C caller and it goes out of scope.
+ // The C code cannot hold a smart pointer so we increment the reference
+ // count to indicate that the C app owns a reference.
+ audioStream->incStrong(nullptr);
+ return audioStream.get();
+}
+
+void AudioStreamBuilder::stopUsingStream(AudioStream *stream) {
+ // Undo the effect of startUsingStream()
+ android::sp<AudioStream> spAudioStream(stream);
+ ALOGV("%s() strongCount = %d", __func__, spAudioStream->getStrongCount());
+ spAudioStream->decStrong(nullptr);
+}
+
aaudio_result_t AudioStreamBuilder::validate() const {
// Check for values that are ridiculously out of range to prevent math overflow exploits.
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.h b/media/libaaudio/src/core/AudioStreamBuilder.h
index d5fb80d..9f93341 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.h
+++ b/media/libaaudio/src/core/AudioStreamBuilder.h
@@ -108,9 +108,16 @@
virtual aaudio_result_t validate() const override;
+
void logParameters() const;
+ // Mark the stream so it can be deleted.
+ static void stopUsingStream(AudioStream *stream);
+
private:
+ // Extract a raw pointer that we can pass to a 'C' app.
+ static AudioStream *startUsingStream(android::sp<AudioStream> &spAudioStream);
+
bool mSharingModeMatchRequired = false; // must match sharing mode requested
aaudio_performance_mode_t mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index c062882..33c1bf5 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -34,8 +34,7 @@
using namespace aaudio;
AudioStreamLegacy::AudioStreamLegacy()
- : AudioStream()
- , mDeviceCallback(new StreamDeviceCallback(this)) {
+ : AudioStream() {
}
AudioStreamLegacy::~AudioStreamLegacy() {
@@ -163,7 +162,11 @@
}
void AudioStreamLegacy::forceDisconnect(bool errorCallbackEnabled) {
- if (getState() != AAUDIO_STREAM_STATE_DISCONNECTED) {
+ // There is no need to disconnect if already in these states.
+ if (getState() != AAUDIO_STREAM_STATE_DISCONNECTED
+ && getState() != AAUDIO_STREAM_STATE_CLOSING
+ && getState() != AAUDIO_STREAM_STATE_CLOSED
+ ) {
setState(AAUDIO_STREAM_STATE_DISCONNECTED);
if (errorCallbackEnabled) {
maybeCallErrorCallback(AAUDIO_ERROR_DISCONNECTED);
@@ -205,24 +208,30 @@
return AAudioConvert_androidToAAudioResult(status);
}
-void AudioStreamLegacy::onAudioDeviceUpdate(audio_port_handle_t deviceId)
-{
+void AudioStreamLegacy::onAudioDeviceUpdate(audio_io_handle_t /* audioIo */,
+ audio_port_handle_t deviceId) {
// Device routing is a common source of errors and DISCONNECTS.
- // Please leave this log in place.
- ALOGD("%s() devId %d => %d", __func__, (int) getDeviceId(), (int)deviceId);
- if (getDeviceId() != AAUDIO_UNSPECIFIED && getDeviceId() != deviceId &&
- getState() != AAUDIO_STREAM_STATE_DISCONNECTED) {
+ // Please leave this log in place. If there is a bug then this might
+ // get called after the stream has been deleted so log before we
+ // touch the stream object.
+ ALOGD("%s(deviceId = %d)", __func__, (int)deviceId);
+ if (getDeviceId() != AAUDIO_UNSPECIFIED
+ && getDeviceId() != deviceId
+ && getState() != AAUDIO_STREAM_STATE_DISCONNECTED
+ ) {
// Note that isDataCallbackActive() is affected by state so call it before DISCONNECTING.
// If we have a data callback and the stream is active, then ask the data callback
// to DISCONNECT and call the error callback.
if (isDataCallbackActive()) {
- ALOGD("onAudioDeviceUpdate() request DISCONNECT in data callback due to device change");
+ ALOGD("%s() request DISCONNECT in data callback, device %d => %d",
+ __func__, (int) getDeviceId(), (int) deviceId);
// If the stream is stopped before the data callback has a chance to handle the
// request then the requestStop() and requestPause() methods will handle it after
// the callback has stopped.
mRequestDisconnect.request();
} else {
- ALOGD("onAudioDeviceUpdate() DISCONNECT the stream now");
+ ALOGD("%s() DISCONNECT the stream now, device %d => %d",
+ __func__, (int) getDeviceId(), (int) deviceId);
forceDisconnect();
}
}
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index 9c24b2b..fefe6e0 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -87,29 +87,13 @@
protected:
- class StreamDeviceCallback : public android::AudioSystem::AudioDeviceCallback
- {
- public:
-
- StreamDeviceCallback(AudioStreamLegacy *parent) : mParent(parent) {}
- virtual ~StreamDeviceCallback() {}
-
- virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo __unused,
- audio_port_handle_t deviceId) {
- if (mParent != nullptr) {
- mParent->onAudioDeviceUpdate(deviceId);
- }
- }
-
- AudioStreamLegacy *mParent;
- };
-
aaudio_result_t getBestTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds,
android::ExtendedTimestamp *extendedTimestamp);
- void onAudioDeviceUpdate(audio_port_handle_t deviceId);
+ void onAudioDeviceUpdate(audio_io_handle_t audioIo,
+ audio_port_handle_t deviceId) override;
/*
* Check to see whether a callback thread has requested a disconnected.
@@ -140,7 +124,6 @@
int32_t mBlockAdapterBytesPerFrame = 0;
aaudio_wrapping_frames_t mPositionWhenStarting = 0;
int32_t mCallbackBufferSize = 0;
- const android::sp<StreamDeviceCallback> mDeviceCallback;
AtomicRequestor mRequestDisconnect;
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index b0dc59e..d62951e 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -282,7 +282,7 @@
: (aaudio_session_id_t) mAudioRecord->getSessionId();
setSessionId(actualSessionId);
- mAudioRecord->addAudioDeviceCallback(mDeviceCallback);
+ mAudioRecord->addAudioDeviceCallback(this);
return AAUDIO_OK;
}
@@ -291,16 +291,24 @@
// TODO add close() or release() to AudioFlinger's AudioRecord API.
// Then call it from here
if (getState() != AAUDIO_STREAM_STATE_CLOSING) {
- mAudioRecord->removeAudioDeviceCallback(mDeviceCallback);
+ mAudioRecord->removeAudioDeviceCallback(this);
logReleaseBufferState();
- mAudioRecord.clear();
- mFixedBlockWriter.close();
+ // Data callbacks may still be running!
return AudioStream::release_l();
} else {
return AAUDIO_OK; // already released
}
}
+void AudioStreamRecord::close_l() {
+ mAudioRecord.clear();
+ // Do not close mFixedBlockWriter because a data callback
+ // thread might still be running if someone else has a reference
+ // to mAudioRecord.
+ // It has a unique_ptr to its buffer so it will clean up by itself.
+ AudioStream::close_l();
+}
+
const void * AudioStreamRecord::maybeConvertDeviceData(const void *audioData, int32_t numFrames) {
if (mFormatConversionBufferFloat.get() != nullptr) {
LOG_ALWAYS_FATAL_IF(numFrames > mFormatConversionBufferSizeInFrames,
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index c5944c7..e4ef1c0 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -39,6 +39,7 @@
aaudio_result_t open(const AudioStreamBuilder & builder) override;
aaudio_result_t release_l() override;
+ void close_l() override;
aaudio_result_t requestStart() override;
aaudio_result_t requestStop() override;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 4869480..3831046 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -224,7 +224,7 @@
mInitialBufferCapacity = getBufferCapacity();
mInitialFramesPerBurst = getFramesPerBurst();
- mAudioTrack->addAudioDeviceCallback(mDeviceCallback);
+ mAudioTrack->addAudioDeviceCallback(this);
// Update performance mode based on the actual stream flags.
// For example, if the sample rate is not allowed then you won't get a FAST track.
@@ -253,19 +253,26 @@
aaudio_result_t AudioStreamTrack::release_l() {
if (getState() != AAUDIO_STREAM_STATE_CLOSING) {
- mAudioTrack->removeAudioDeviceCallback(mDeviceCallback);
+ status_t err = mAudioTrack->removeAudioDeviceCallback(this);
+ ALOGE_IF(err, "%s() removeAudioDeviceCallback returned %d", __func__, err);
logReleaseBufferState();
- // TODO Investigate why clear() causes a hang in test_various.cpp
- // if I call close() from a data callback.
- // But the same thing in AudioRecord is OK!
- // mAudioTrack.clear();
- mFixedBlockReader.close();
+ // Data callbacks may still be running!
return AudioStream::release_l();
} else {
return AAUDIO_OK; // already released
}
}
+void AudioStreamTrack::close_l() {
+ // Stop callbacks before deleting mFixedBlockReader memory.
+ mAudioTrack.clear();
+ // Do not close mFixedBlockReader because a data callback
+ // thread might still be running if someone else has a reference
+ // to mAudioRecord.
+ // It has a unique_ptr to its buffer so it will clean up by itself.
+ AudioStream::close_l();
+}
+
void AudioStreamTrack::processCallback(int event, void *info) {
switch (event) {
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index 93a1ff4..6334f66 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h
@@ -42,6 +42,7 @@
aaudio_result_t open(const AudioStreamBuilder & builder) override;
aaudio_result_t release_l() override;
+ void close_l() override;
aaudio_result_t requestStart() override;
aaudio_result_t requestPause() override;
diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
index a20c799..cbf863f 100644
--- a/media/libaaudio/tests/test_various.cpp
+++ b/media/libaaudio/tests/test_various.cpp
@@ -33,6 +33,11 @@
void *audioData,
int32_t numFrames
) {
+ aaudio_direction_t direction = AAudioStream_getDirection(stream);
+ if (direction == AAUDIO_DIRECTION_INPUT) {
+ return AAUDIO_CALLBACK_RESULT_CONTINUE;
+ }
+ // Check to make sure the buffer is initialized to all zeros.
int channels = AAudioStream_getChannelCount(stream);
int numSamples = channels * numFrames;
bool allZeros = true;
@@ -48,7 +53,8 @@
constexpr int64_t NANOS_PER_MILLISECOND = 1000 * 1000;
void checkReleaseThenClose(aaudio_performance_mode_t perfMode,
- aaudio_sharing_mode_t sharingMode) {
+ aaudio_sharing_mode_t sharingMode,
+ aaudio_direction_t direction = AAUDIO_DIRECTION_OUTPUT) {
AAudioStreamBuilder* aaudioBuilder = nullptr;
AAudioStream* aaudioStream = nullptr;
@@ -61,6 +67,7 @@
nullptr);
AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
AAudioStreamBuilder_setSharingMode(aaudioBuilder, sharingMode);
+ AAudioStreamBuilder_setDirection(aaudioBuilder, direction);
AAudioStreamBuilder_setFormat(aaudioBuilder, AAUDIO_FORMAT_PCM_FLOAT);
// Create an AAudioStream using the Builder.
@@ -88,14 +95,28 @@
// We should NOT be able to start or change a stream after it has been released.
EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestStart(aaudioStream));
EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
- EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestPause(aaudioStream));
+ // Pause is only implemented for OUTPUT.
+ if (direction == AAUDIO_DIRECTION_OUTPUT) {
+ EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE,
+ AAudioStream_requestPause(aaudioStream));
+ }
EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestStop(aaudioStream));
EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
// Does this crash?
- EXPECT_LT(0, AAudioStream_getFramesRead(aaudioStream));
- EXPECT_LT(0, AAudioStream_getFramesWritten(aaudioStream));
+ EXPECT_GT(AAudioStream_getFramesRead(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getFramesWritten(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getFramesPerBurst(aaudioStream), 0);
+ EXPECT_GE(AAudioStream_getXRunCount(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getBufferCapacityInFrames(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getBufferSizeInFrames(aaudioStream), 0);
+
+ int64_t timestampFrames = 0;
+ int64_t timestampNanos = 0;
+ aaudio_result_t result = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
+ ×tampFrames, ×tampNanos);
+ EXPECT_TRUE(result == AAUDIO_ERROR_INVALID_STATE || result == AAUDIO_ERROR_UNIMPLEMENTED);
// Verify Closing State. Does this crash?
aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
@@ -107,20 +128,42 @@
EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
}
-TEST(test_various, aaudio_release_close_none) {
+TEST(test_various, aaudio_release_close_none_output) {
checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_NONE,
- AAUDIO_SHARING_MODE_SHARED);
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_OUTPUT);
// No EXCLUSIVE streams with MODE_NONE.
}
-TEST(test_various, aaudio_release_close_low_shared) {
- checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
- AAUDIO_SHARING_MODE_SHARED);
+TEST(test_various, aaudio_release_close_none_input) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_NONE,
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_INPUT);
+ // No EXCLUSIVE streams with MODE_NONE.
}
-TEST(test_various, aaudio_release_close_low_exclusive) {
+TEST(test_various, aaudio_release_close_low_shared_output) {
checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
- AAUDIO_SHARING_MODE_EXCLUSIVE);
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_OUTPUT);
+}
+
+TEST(test_various, aaudio_release_close_low_shared_input) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_INPUT);
+}
+
+TEST(test_various, aaudio_release_close_low_exclusive_output) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_SHARING_MODE_EXCLUSIVE,
+ AAUDIO_DIRECTION_OUTPUT);
+}
+
+TEST(test_various, aaudio_release_close_low_exclusive_input) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_SHARING_MODE_EXCLUSIVE,
+ AAUDIO_DIRECTION_INPUT);
}
enum FunctionToCall {
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 32129f0..68d11d4 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -210,7 +210,11 @@
return NO_ERROR;
}
-AudioTrack::AudioTrack()
+AudioTrack::AudioTrack() : AudioTrack("" /*opPackageName*/)
+{
+}
+
+AudioTrack::AudioTrack(const std::string& opPackageName)
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -218,6 +222,7 @@
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mOpPackageName(opPackageName),
mAudioTrackCallback(new AudioTrackCallback())
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
@@ -244,12 +249,14 @@
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed,
- audio_port_handle_t selectedDeviceId)
+ audio_port_handle_t selectedDeviceId,
+ const std::string& opPackageName)
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
+ mOpPackageName(opPackageName),
mAudioTrackCallback(new AudioTrackCallback())
{
mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
@@ -277,13 +284,15 @@
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
- float maxRequiredSpeed)
+ float maxRequiredSpeed,
+ const std::string& opPackageName)
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mOpPackageName(opPackageName),
mAudioTrackCallback(new AudioTrackCallback())
{
mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
@@ -1555,6 +1564,7 @@
input.selectedDeviceId = mSelectedDeviceId;
input.sessionId = mSessionId;
input.audioTrackCallback = mAudioTrackCallback;
+ input.opPackageName = mOpPackageName;
IAudioFlinger::CreateTrackOutput output;
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 0dbd842..a9946da 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -26,6 +26,8 @@
#include <media/Modulo.h>
#include <utils/threads.h>
+#include <string>
+
#include "android/media/BnAudioTrackCallback.h"
#include "android/media/IAudioTrackCallback.h"
@@ -177,6 +179,8 @@
*/
AudioTrack();
+ AudioTrack(const std::string& opPackageName);
+
/* Creates an AudioTrack object and registers it with AudioFlinger.
* Once created, the track needs to be started before it can be used.
* Unspecified values are set to appropriate default values.
@@ -258,7 +262,8 @@
const audio_attributes_t* pAttributes = NULL,
bool doNotReconnect = false,
float maxRequiredSpeed = 1.0f,
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
+ const std::string& opPackageName = "");
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
@@ -288,7 +293,8 @@
pid_t pid = -1,
const audio_attributes_t* pAttributes = NULL,
bool doNotReconnect = false,
- float maxRequiredSpeed = 1.0f);
+ float maxRequiredSpeed = 1.0f,
+ const std::string& opPackageName = "");
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
@@ -1236,6 +1242,8 @@
sp<media::VolumeHandler> mVolumeHandler;
+ const std::string mOpPackageName;
+
private:
class DeathNotifier : public IBinder::DeathRecipient {
public:
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 612ce7a..bcc11f4 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -37,6 +37,7 @@
#include <media/IEffectClient.h>
#include <utils/String8.h>
#include <media/MicrophoneInfo.h>
+#include <string>
#include <vector>
#include "android/media/IAudioRecord.h"
@@ -85,6 +86,11 @@
speed = parcel->readFloat();
audioTrackCallback = interface_cast<media::IAudioTrackCallback>(
parcel->readStrongBinder());
+ const char* opPackageNamePtr = parcel->readCString();
+ if (opPackageNamePtr == nullptr) {
+ return FAILED_TRANSACTION;
+ }
+ opPackageName = opPackageNamePtr;
/* input/output arguments*/
(void)parcel->read(&flags, sizeof(audio_output_flags_t));
@@ -109,6 +115,7 @@
(void)parcel->writeInt32(notificationsPerBuffer);
(void)parcel->writeFloat(speed);
(void)parcel->writeStrongBinder(IInterface::asBinder(audioTrackCallback));
+ (void)parcel->writeCString(opPackageName.c_str());
/* input/output arguments*/
(void)parcel->write(&flags, sizeof(audio_output_flags_t));
@@ -127,6 +134,7 @@
uint32_t notificationsPerBuffer;
float speed;
sp<media::IAudioTrackCallback> audioTrackCallback;
+ std::string opPackageName;
/* input/output */
audio_output_flags_t flags;
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index bd18a40..11005c6 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -62,11 +62,13 @@
}
virtual sp<IMediaPlayer> create(
- const sp<IMediaPlayerClient>& client, audio_session_t audioSessionId) {
+ const sp<IMediaPlayerClient>& client, audio_session_t audioSessionId,
+ const std::string opPackageName) {
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
data.writeStrongBinder(IInterface::asBinder(client));
data.writeInt32(audioSessionId);
+ data.writeCString(opPackageName.c_str());
remote()->transact(CREATE, data, &reply);
return interface_cast<IMediaPlayer>(reply.readStrongBinder());
@@ -127,7 +129,12 @@
sp<IMediaPlayerClient> client =
interface_cast<IMediaPlayerClient>(data.readStrongBinder());
audio_session_t audioSessionId = (audio_session_t) data.readInt32();
- sp<IMediaPlayer> player = create(client, audioSessionId);
+ const char* opPackageName = data.readCString();
+ if (opPackageName == nullptr) {
+ return FAILED_TRANSACTION;
+ }
+ std::string opPackageNameStr(opPackageName);
+ sp<IMediaPlayer> player = create(client, audioSessionId, opPackageNameStr);
reply->writeStrongBinder(IInterface::asBinder(player));
return NO_ERROR;
} break;
diff --git a/media/libmedia/include/media/IMediaPlayerService.h b/media/libmedia/include/media/IMediaPlayerService.h
index f2e2060..a4207eb 100644
--- a/media/libmedia/include/media/IMediaPlayerService.h
+++ b/media/libmedia/include/media/IMediaPlayerService.h
@@ -28,6 +28,8 @@
#include <media/IMediaPlayerClient.h>
#include <media/IMediaMetadataRetriever.h>
+#include <string>
+
namespace android {
class IMediaPlayer;
@@ -47,7 +49,8 @@
virtual sp<IMediaRecorder> createMediaRecorder(const String16 &opPackageName) = 0;
virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client,
- audio_session_t audioSessionId = AUDIO_SESSION_ALLOCATE) = 0;
+ audio_session_t audioSessionId = AUDIO_SESSION_ALLOCATE,
+ const std::string opPackage = "") = 0;
virtual sp<IMediaCodecList> getCodecList() const = 0;
// Connects to a remote display.
diff --git a/media/libmedia/include/media/mediaplayer.h b/media/libmedia/include/media/mediaplayer.h
index 2335c5a..7c29e50 100644
--- a/media/libmedia/include/media/mediaplayer.h
+++ b/media/libmedia/include/media/mediaplayer.h
@@ -33,6 +33,8 @@
#include <utils/KeyedVector.h>
#include <utils/String8.h>
+#include <string>
+
struct ANativeWindow;
namespace android {
@@ -205,7 +207,7 @@
public virtual IMediaDeathNotifier
{
public:
- MediaPlayer();
+ MediaPlayer(const std::string opPackageName = "");
~MediaPlayer();
void died();
void disconnect();
@@ -308,6 +310,7 @@
float mSendLevel;
struct sockaddr_in mRetransmitEndpoint;
bool mRetransmitEndpointValid;
+ const std::string mOpPackageName;
};
}; // namespace android
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 1fadc94..6079a2d 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -41,7 +41,7 @@
using media::VolumeShaper;
-MediaPlayer::MediaPlayer()
+MediaPlayer::MediaPlayer(const std::string opPackageName) : mOpPackageName(opPackageName)
{
ALOGV("constructor");
mListener = NULL;
@@ -152,7 +152,7 @@
if (url != NULL) {
const sp<IMediaPlayerService> service(getMediaPlayerService());
if (service != 0) {
- sp<IMediaPlayer> player(service->create(this, mAudioSessionId));
+ sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mOpPackageName));
if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
(NO_ERROR != player->setDataSource(httpService, url, headers))) {
player.clear();
@@ -169,7 +169,7 @@
status_t err = UNKNOWN_ERROR;
const sp<IMediaPlayerService> service(getMediaPlayerService());
if (service != 0) {
- sp<IMediaPlayer> player(service->create(this, mAudioSessionId));
+ sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mOpPackageName));
if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
(NO_ERROR != player->setDataSource(fd, offset, length))) {
player.clear();
@@ -185,7 +185,7 @@
status_t err = UNKNOWN_ERROR;
const sp<IMediaPlayerService> service(getMediaPlayerService());
if (service != 0) {
- sp<IMediaPlayer> player(service->create(this, mAudioSessionId));
+ sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mOpPackageName));
if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
(NO_ERROR != player->setDataSource(source))) {
player.clear();
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index c0da0ce..016f622 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -480,14 +480,14 @@
}
sp<IMediaPlayer> MediaPlayerService::create(const sp<IMediaPlayerClient>& client,
- audio_session_t audioSessionId)
+ audio_session_t audioSessionId, std::string opPackageName)
{
pid_t pid = IPCThreadState::self()->getCallingPid();
int32_t connId = android_atomic_inc(&mNextConnId);
sp<Client> c = new Client(
this, pid, connId, client, audioSessionId,
- IPCThreadState::self()->getCallingUid());
+ IPCThreadState::self()->getCallingUid(), opPackageName);
ALOGV("Create new client(%d) from pid %d, uid %d, ", connId, pid,
IPCThreadState::self()->getCallingUid());
@@ -733,7 +733,8 @@
MediaPlayerService::Client::Client(
const sp<MediaPlayerService>& service, pid_t pid,
int32_t connId, const sp<IMediaPlayerClient>& client,
- audio_session_t audioSessionId, uid_t uid)
+ audio_session_t audioSessionId, uid_t uid, const std::string& opPackageName)
+ : mOpPackageName(opPackageName)
{
ALOGV("Client(%d) constructor", connId);
mPid = pid;
@@ -922,7 +923,7 @@
if (!p->hardwareOutput()) {
mAudioOutput = new AudioOutput(mAudioSessionId, IPCThreadState::self()->getCallingUid(),
- mPid, mAudioAttributes, mAudioDeviceUpdatedListener);
+ mPid, mAudioAttributes, mAudioDeviceUpdatedListener, mOpPackageName);
static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
}
@@ -1761,7 +1762,8 @@
#undef LOG_TAG
#define LOG_TAG "AudioSink"
MediaPlayerService::AudioOutput::AudioOutput(audio_session_t sessionId, uid_t uid, int pid,
- const audio_attributes_t* attr, const sp<AudioSystem::AudioDeviceCallback>& deviceCallback)
+ const audio_attributes_t* attr, const sp<AudioSystem::AudioDeviceCallback>& deviceCallback,
+ const std::string& opPackageName)
: mCallback(NULL),
mCallbackCookie(NULL),
mCallbackData(NULL),
@@ -1782,7 +1784,8 @@
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
mDeviceCallbackEnabled(false),
- mDeviceCallback(deviceCallback)
+ mDeviceCallback(deviceCallback),
+ mOpPackageName(opPackageName)
{
ALOGV("AudioOutput(%d)", sessionId);
if (attr != NULL) {
@@ -2176,7 +2179,8 @@
mAttributes,
doNotReconnect,
1.0f, // default value for maxRequiredSpeed
- mSelectedDeviceId);
+ mSelectedDeviceId,
+ mOpPackageName);
} else {
// TODO: Due to buffer memory concerns, we use a max target playback speed
// based on mPlaybackRate at the time of open (instead of kMaxRequiredSpeed),
@@ -2204,7 +2208,8 @@
mAttributes,
doNotReconnect,
targetSpeed,
- mSelectedDeviceId);
+ mSelectedDeviceId,
+ mOpPackageName);
}
// Set caller name so it can be logged in destructor.
// MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_MEDIA
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 6431ca1..a7de3f3 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -19,6 +19,7 @@
#define ANDROID_MEDIAPLAYERSERVICE_H
#include <arpa/inet.h>
+#include <string>
#include <utils/threads.h>
#include <utils/Errors.h>
@@ -81,7 +82,8 @@
uid_t uid,
int pid,
const audio_attributes_t * attr,
- const sp<AudioSystem::AudioDeviceCallback>& deviceCallback);
+ const sp<AudioSystem::AudioDeviceCallback>& deviceCallback,
+ const std::string& opPackageName);
virtual ~AudioOutput();
virtual bool ready() const { return mTrack != 0; }
@@ -178,6 +180,7 @@
bool mDeviceCallbackEnabled;
wp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
mutable Mutex mLock;
+ const std::string mOpPackageName;
// static variables below not protected by mutex
static bool mIsOnEmulator;
@@ -235,7 +238,8 @@
virtual sp<IMediaMetadataRetriever> createMetadataRetriever();
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client,
- audio_session_t audioSessionId);
+ audio_session_t audioSessionId,
+ const std::string opPackageName);
virtual sp<IMediaCodecList> getCodecList() const;
@@ -410,7 +414,8 @@
int32_t connId,
const sp<IMediaPlayerClient>& client,
audio_session_t audioSessionId,
- uid_t uid);
+ uid_t uid,
+ const std::string& opPackageName);
Client();
virtual ~Client();
@@ -467,6 +472,7 @@
bool mRetransmitEndpointValid;
sp<Client> mNextClient;
sp<MediaPlayerBase::Listener> mListener;
+ const std::string mOpPackageName;
// Metadata filters.
media::Metadata::Filter mMetadataAllow; // protected by mLock
diff --git a/media/utils/ServiceUtilities.cpp b/media/utils/ServiceUtilities.cpp
index 87ea084..7699700 100644
--- a/media/utils/ServiceUtilities.cpp
+++ b/media/utils/ServiceUtilities.cpp
@@ -62,7 +62,7 @@
}
static bool checkRecordingInternal(const String16& opPackageName, pid_t pid,
- uid_t uid, bool start) {
+ uid_t uid, bool start, bool isHotwordSource) {
// Okay to not track in app ops as audio server or media server is us and if
// device is rooted security model is considered compromised.
// system_server loses its RECORD_AUDIO permission when a secondary
@@ -87,16 +87,21 @@
}
AppOpsManager appOps;
- const int32_t op = appOps.permissionToOpCode(sAndroidPermissionRecordAudio);
+ const int32_t opRecordAudio = appOps.permissionToOpCode(sAndroidPermissionRecordAudio);
+
if (start) {
+ const int32_t op = isHotwordSource ?
+ AppOpsManager::OP_RECORD_AUDIO_HOTWORD : opRecordAudio;
if (appOps.startOpNoThrow(op, uid, resolvedOpPackageName, /*startIfModeDefault*/ false)
!= AppOpsManager::MODE_ALLOWED) {
ALOGE("Request denied by app op: %d", op);
return false;
}
} else {
- if (appOps.checkOp(op, uid, resolvedOpPackageName) != AppOpsManager::MODE_ALLOWED) {
- ALOGE("Request denied by app op: %d", op);
+ // Always use OP_RECORD_AUDIO for checks at creation time.
+ if (appOps.checkOp(opRecordAudio, uid, resolvedOpPackageName)
+ != AppOpsManager::MODE_ALLOWED) {
+ ALOGE("Request denied by app op: %d", opRecordAudio);
return false;
}
}
@@ -105,14 +110,15 @@
}
bool recordingAllowed(const String16& opPackageName, pid_t pid, uid_t uid) {
- return checkRecordingInternal(opPackageName, pid, uid, /*start*/ false);
+ return checkRecordingInternal(opPackageName, pid, uid, /*start*/ false,
+ /*is_hotword_source*/ false);
}
-bool startRecording(const String16& opPackageName, pid_t pid, uid_t uid) {
- return checkRecordingInternal(opPackageName, pid, uid, /*start*/ true);
+bool startRecording(const String16& opPackageName, pid_t pid, uid_t uid, bool isHotwordSource) {
+ return checkRecordingInternal(opPackageName, pid, uid, /*start*/ true, isHotwordSource);
}
-void finishRecording(const String16& opPackageName, uid_t uid) {
+void finishRecording(const String16& opPackageName, uid_t uid, bool isHotwordSource) {
// Okay to not track in app ops as audio server is us and if
// device is rooted security model is considered compromised.
if (isAudioServerOrRootUid(uid)) return;
@@ -125,7 +131,8 @@
}
AppOpsManager appOps;
- const int32_t op = appOps.permissionToOpCode(sAndroidPermissionRecordAudio);
+ const int32_t op = isHotwordSource ? AppOpsManager::OP_RECORD_AUDIO_HOTWORD
+ : appOps.permissionToOpCode(sAndroidPermissionRecordAudio);
appOps.finishOp(op, uid, resolvedOpPackageName);
}
diff --git a/media/utils/include/mediautils/ServiceUtilities.h b/media/utils/include/mediautils/ServiceUtilities.h
index 212599a..431dd7a 100644
--- a/media/utils/include/mediautils/ServiceUtilities.h
+++ b/media/utils/include/mediautils/ServiceUtilities.h
@@ -79,8 +79,8 @@
}
bool recordingAllowed(const String16& opPackageName, pid_t pid, uid_t uid);
-bool startRecording(const String16& opPackageName, pid_t pid, uid_t uid);
-void finishRecording(const String16& opPackageName, uid_t uid);
+bool startRecording(const String16& opPackageName, pid_t pid, uid_t uid, bool isHotwordSource);
+void finishRecording(const String16& opPackageName, uid_t uid, bool isHotwordSource);
bool captureAudioOutputAllowed(pid_t pid, uid_t uid);
bool captureMediaOutputAllowed(pid_t pid, uid_t uid);
bool captureVoiceCommunicationOutputAllowed(pid_t pid, uid_t uid);
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2ef64d6..764fdc3 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -852,7 +852,8 @@
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
callingPid, input.clientInfo.clientTid, clientUid,
- &lStatus, portId, input.audioTrackCallback);
+ &lStatus, portId, input.audioTrackCallback,
+ input.opPackageName);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index d8eebf3..a2df29b 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -26,10 +26,11 @@
bool hasOpPlayAudio() const;
static sp<OpPlayAudioMonitor> createIfNeeded(
- uid_t uid, const audio_attributes_t& attr, int id, audio_stream_type_t streamType);
+ uid_t uid, const audio_attributes_t& attr, int id, audio_stream_type_t streamType,
+ const std::string& opPackageName);
private:
- OpPlayAudioMonitor(uid_t uid, audio_usage_t usage, int id);
+ OpPlayAudioMonitor(uid_t uid, audio_usage_t usage, int id, const String16& opPackageName);
void onFirstRef() override;
static void getPackagesForUid(uid_t uid, Vector<String16>& packages);
@@ -49,10 +50,10 @@
void checkPlayAudioForUsage();
std::atomic_bool mHasOpPlayAudio;
- Vector<String16> mPackages;
const uid_t mUid;
const int32_t mUsage; // on purpose not audio_usage_t because always checked in appOps as int32_t
const int mId; // for logging purposes only
+ const String16 mOpPackageName;
};
// playback track
@@ -77,7 +78,8 @@
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
/** default behaviour is to start when there are as many frames
* ready as possible (aka. Buffer is full). */
- size_t frameCountToBeReady = SIZE_MAX);
+ size_t frameCountToBeReady = SIZE_MAX,
+ const std::string opPackageName = "");
virtual ~Track();
virtual status_t initCheck() const;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 5279bb3..766f940 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2067,7 +2067,8 @@
uid_t uid,
status_t *status,
audio_port_handle_t portId,
- const sp<media::IAudioTrackCallback>& callback)
+ const sp<media::IAudioTrackCallback>& callback,
+ const std::string& opPackageName)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
@@ -2348,7 +2349,8 @@
track = new Track(this, client, streamType, attr, sampleRate, format,
channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
- sessionId, creatorPid, uid, *flags, TrackBase::TYPE_DEFAULT, portId);
+ sessionId, creatorPid, uid, *flags, TrackBase::TYPE_DEFAULT, portId,
+ SIZE_MAX /*frameCountToBeReady*/, opPackageName);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index c1ac2e4..f95e0b8 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -864,7 +864,8 @@
uid_t uid,
status_t *status /*non-NULL*/,
audio_port_handle_t portId,
- const sp<media::IAudioTrackCallback>& callback);
+ const sp<media::IAudioTrackCallback>& callback,
+ const std::string& opPackageName);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 8947106..f286d8a 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -386,11 +386,12 @@
// static
sp<AudioFlinger::PlaybackThread::OpPlayAudioMonitor>
AudioFlinger::PlaybackThread::OpPlayAudioMonitor::createIfNeeded(
- uid_t uid, const audio_attributes_t& attr, int id, audio_stream_type_t streamType)
+ uid_t uid, const audio_attributes_t& attr, int id, audio_stream_type_t streamType,
+ const std::string& opPackageName)
{
+ Vector <String16> packages;
+ getPackagesForUid(uid, packages);
if (isServiceUid(uid)) {
- Vector <String16> packages;
- getPackagesForUid(uid, packages);
if (packages.isEmpty()) {
ALOGD("OpPlayAudio: not muting track:%d usage:%d for service UID %d",
id,
@@ -410,12 +411,32 @@
id, attr.flags);
return nullptr;
}
- return new OpPlayAudioMonitor(uid, attr.usage, id);
+
+ String16 opPackageNameStr(opPackageName.c_str());
+ if (opPackageName.empty()) {
+ // If no package name is provided by the client, use the first associated with the uid
+ if (!packages.isEmpty()) {
+ opPackageNameStr = packages[0];
+ }
+ } else {
+ // If the provided package name is invalid, we force app ops denial by clearing the package
+ // name passed to OpPlayAudioMonitor
+ if (std::find_if(packages.begin(), packages.end(),
+ [&opPackageNameStr](const auto& package) {
+ return opPackageNameStr == package; }) == packages.end()) {
+ ALOGW("The package name(%s) provided does not correspond to the uid %d, "
+ "force muting the track", opPackageName.c_str(), uid);
+ // Set package name as an empty string so that hasOpPlayAudio will always return false.
+ opPackageNameStr = String16("");
+ }
+ }
+ return new OpPlayAudioMonitor(uid, attr.usage, id, opPackageNameStr);
}
AudioFlinger::PlaybackThread::OpPlayAudioMonitor::OpPlayAudioMonitor(
- uid_t uid, audio_usage_t usage, int id)
- : mHasOpPlayAudio(true), mUid(uid), mUsage((int32_t) usage), mId(id)
+ uid_t uid, audio_usage_t usage, int id, const String16& opPackageName)
+ : mHasOpPlayAudio(true), mUid(uid), mUsage((int32_t) usage), mId(id),
+ mOpPackageName(opPackageName)
{
}
@@ -429,11 +450,10 @@
void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::onFirstRef()
{
- getPackagesForUid(mUid, mPackages);
checkPlayAudioForUsage();
- if (!mPackages.isEmpty()) {
+ if (mOpPackageName.size() != 0) {
mOpCallback = new PlayAudioOpCallback(this);
- mAppOpsManager.startWatchingMode(AppOpsManager::OP_PLAY_AUDIO, mPackages[0], mOpCallback);
+ mAppOpsManager.startWatchingMode(AppOpsManager::OP_PLAY_AUDIO, mOpPackageName, mOpCallback);
}
}
@@ -446,18 +466,11 @@
// - not called from PlayAudioOpCallback because the callback is not installed in this case
void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::checkPlayAudioForUsage()
{
- if (mPackages.isEmpty()) {
+ if (mOpPackageName.size() == 0) {
mHasOpPlayAudio.store(false);
} else {
- bool hasIt = true;
- for (const String16& packageName : mPackages) {
- const int32_t mode = mAppOpsManager.checkAudioOpNoThrow(AppOpsManager::OP_PLAY_AUDIO,
- mUsage, mUid, packageName);
- if (mode != AppOpsManager::MODE_ALLOWED) {
- hasIt = false;
- break;
- }
- }
+ bool hasIt = mAppOpsManager.checkAudioOpNoThrow(AppOpsManager::OP_PLAY_AUDIO,
+ mUsage, mUid, mOpPackageName) == AppOpsManager::MODE_ALLOWED;
ALOGD("OpPlayAudio: track:%d usage:%d %smuted", mId, mUsage, hasIt ? "not " : "");
mHasOpPlayAudio.store(hasIt);
}
@@ -511,7 +524,8 @@
audio_output_flags_t flags,
track_type type,
audio_port_handle_t portId,
- size_t frameCountToBeReady)
+ size_t frameCountToBeReady,
+ const std::string opPackageName)
: TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
@@ -534,7 +548,8 @@
mPresentationCompleteFrames(0),
mFrameMap(16 /* sink-frame-to-track-frame map memory */),
mVolumeHandler(new media::VolumeHandler(sampleRate)),
- mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(uid, attr, id(), streamType)),
+ mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(
+ uid, attr, id(), streamType, opPackageName)),
// mSinkTimestamp
mFrameCountToBeReady(frameCountToBeReady),
mFastIndex(-1),
@@ -598,7 +613,7 @@
if (channelMask & AUDIO_CHANNEL_HAPTIC_ALL) {
mAudioVibrationController = new AudioVibrationController(this);
mExternalVibration = new os::ExternalVibration(
- mUid, "" /* pkg */, mAttr, mAudioVibrationController);
+ mUid, opPackageName, mAttr, mAudioVibrationController);
}
// Once this item is logged by the server, the client can add properties.
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 6522016..07e80d1 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -3161,6 +3161,8 @@
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
waitMs = updateCallRouting(newDevices, delayMs);
+ // Only apply special touch sound delay once
+ delayMs = 0;
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
@@ -3170,6 +3172,8 @@
// preventing the force re-routing in case of default dev that distinguishes on address.
// Let's give back to engine full device choice decision however.
waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
+ // Only apply special touch sound delay once
+ delayMs = 0;
}
if (forceVolumeReeval && !newDevices.isEmpty()) {
applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
@@ -5648,7 +5652,7 @@
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
setVolumeSourceMute(ttsVolumeSource, mute/*on*/, desc, 0 /*delay*/, DeviceTypeSet());
const uint32_t latency = desc->latency() * 2;
- if (latency > maxLatency) {
+ if (desc->isActive(latency * 2) && latency > maxLatency) {
maxLatency = latency;
}
}
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 34d07b6..df27f6e 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -572,7 +572,8 @@
}
// check calling permissions
- if (!(startRecording(client->opPackageName, client->pid, client->uid)
+ if (!(startRecording(client->opPackageName, client->pid, client->uid,
+ client->attributes.source == AUDIO_SOURCE_HOTWORD)
|| client->attributes.source == AUDIO_SOURCE_FM_TUNER)) {
ALOGE("%s permission denied: recording not allowed for uid %d pid %d",
__func__, client->uid, client->pid);
@@ -660,7 +661,8 @@
client->active = false;
client->startTimeNs = 0;
updateUidStates_l();
- finishRecording(client->opPackageName, client->uid);
+ finishRecording(client->opPackageName, client->uid,
+ client->attributes.source == AUDIO_SOURCE_HOTWORD);
}
return status;
@@ -686,7 +688,8 @@
updateUidStates_l();
// finish the recording app op
- finishRecording(client->opPackageName, client->uid);
+ finishRecording(client->opPackageName, client->uid,
+ client->attributes.source == AUDIO_SOURCE_HOTWORD);
AutoCallerClear acc;
return mAudioPolicyManager->stopInput(portId);
}
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index af1e01d..24288d6 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -3751,9 +3751,14 @@
__FUNCTION__, cameraId.string());
return;
}
+
+ // Collect the logical cameras without holding mStatusLock in updateStatus
+ // as that can lead to a deadlock(b/162192331).
+ auto logicalCameraIds = getLogicalCameras(cameraId);
// Update the status for this camera state, then send the onStatusChangedCallbacks to each
// of the listeners with both the mStatusLock and mStatusListenerLock held
- state->updateStatus(status, cameraId, rejectSourceStates, [this, &deviceKind, &supportsHAL3]
+ state->updateStatus(status, cameraId, rejectSourceStates, [this, &deviceKind, &supportsHAL3,
+ &logicalCameraIds]
(const String8& cameraId, StatusInternal status) {
if (status != StatusInternal::ENUMERATING) {
@@ -3773,8 +3778,8 @@
}
Mutex::Autolock lock(mStatusListenerLock);
-
- notifyPhysicalCameraStatusLocked(mapToInterface(status), cameraId, deviceKind);
+ notifyPhysicalCameraStatusLocked(mapToInterface(status), String16(cameraId),
+ logicalCameraIds, deviceKind);
for (auto& listener : mListenerList) {
bool isVendorListener = listener->isVendorListener();
@@ -3892,8 +3897,9 @@
return OK;
}
-void CameraService::notifyPhysicalCameraStatusLocked(int32_t status, const String8& cameraId,
- SystemCameraKind deviceKind) {
+std::list<String16> CameraService::getLogicalCameras(
+ const String8& physicalCameraId) {
+ std::list<String16> retList;
Mutex::Autolock lock(mCameraStatesLock);
for (const auto& state : mCameraStates) {
std::vector<std::string> physicalCameraIds;
@@ -3901,26 +3907,39 @@
// This is not a logical multi-camera.
continue;
}
- if (std::find(physicalCameraIds.begin(), physicalCameraIds.end(), cameraId.c_str())
+ if (std::find(physicalCameraIds.begin(), physicalCameraIds.end(), physicalCameraId.c_str())
== physicalCameraIds.end()) {
// cameraId is not a physical camera of this logical multi-camera.
continue;
}
- String16 id16(state.first), physicalId16(cameraId);
+ retList.emplace_back(String16(state.first));
+ }
+ return retList;
+}
+
+void CameraService::notifyPhysicalCameraStatusLocked(int32_t status,
+ const String16& physicalCameraId, const std::list<String16>& logicalCameraIds,
+ SystemCameraKind deviceKind) {
+ // mStatusListenerLock is expected to be locked
+ for (const auto& logicalCameraId : logicalCameraIds) {
for (auto& listener : mListenerList) {
+ // Note: we check only the deviceKind of the physical camera id
+ // since, logical camera ids and their physical camera ids are
+ // guaranteed to have the same system camera kind.
if (shouldSkipStatusUpdates(deviceKind, listener->isVendorListener(),
listener->getListenerPid(), listener->getListenerUid())) {
ALOGV("Skipping discovery callback for system-only camera device %s",
- cameraId.c_str());
+ String8(physicalCameraId).c_str());
continue;
}
listener->getListener()->onPhysicalCameraStatusChanged(status,
- id16, physicalId16);
+ logicalCameraId, physicalCameraId);
}
}
}
+
void CameraService::blockClientsForUid(uid_t uid) {
const auto clients = mActiveClientManager.getAll();
for (auto& current : clients) {
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 4321201..685ed5e 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -49,6 +49,7 @@
#include <set>
#include <string>
+#include <list>
#include <map>
#include <memory>
#include <utility>
@@ -1005,8 +1006,13 @@
hardware::camera::common::V1_0::TorchModeStatus status);
// notify physical camera status when the physical camera is public.
- void notifyPhysicalCameraStatusLocked(int32_t status, const String8& cameraId,
- SystemCameraKind deviceKind);
+ // Expects mStatusListenerLock to be locked.
+ void notifyPhysicalCameraStatusLocked(int32_t status, const String16& physicalCameraId,
+ const std::list<String16>& logicalCameraIds, SystemCameraKind deviceKind);
+
+ // get list of logical cameras which are backed by physicalCameraId
+ std::list<String16> getLogicalCameras(const String8& physicalCameraId);
+
// IBinder::DeathRecipient implementation
virtual void binderDied(const wp<IBinder> &who);
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index a171cb0..04b906a 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -47,7 +47,11 @@
virtual aaudio_result_t open(const aaudio::AAudioStreamRequest &request) = 0;
- virtual aaudio_result_t close() = 0;
+ /*
+ * Perform any cleanup necessary before deleting the stream.
+ * This might include releasing and closing internal streams.
+ */
+ virtual void close() = 0;
aaudio_result_t registerStream(android::sp<AAudioServiceStreamBase> stream);
diff --git a/services/oboeservice/AAudioServiceEndpointCapture.cpp b/services/oboeservice/AAudioServiceEndpointCapture.cpp
index 37d105b..220584c 100644
--- a/services/oboeservice/AAudioServiceEndpointCapture.cpp
+++ b/services/oboeservice/AAudioServiceEndpointCapture.cpp
@@ -36,18 +36,18 @@
using namespace aaudio; // TODO just import names needed
AAudioServiceEndpointCapture::AAudioServiceEndpointCapture(AAudioService &audioService)
- : mStreamInternalCapture(audioService, true) {
- mStreamInternal = &mStreamInternalCapture;
+ : AAudioServiceEndpointShared(
+ (AudioStreamInternal *)(new AudioStreamInternalCapture(audioService, true))) {
}
AAudioServiceEndpointCapture::~AAudioServiceEndpointCapture() {
- delete mDistributionBuffer;
+ delete[] mDistributionBuffer;
}
aaudio_result_t AAudioServiceEndpointCapture::open(const aaudio::AAudioStreamRequest &request) {
aaudio_result_t result = AAudioServiceEndpointShared::open(request);
if (result == AAUDIO_OK) {
- delete mDistributionBuffer;
+ delete[] mDistributionBuffer;
int distributionBufferSizeBytes = getStreamInternal()->getFramesPerBurst()
* getStreamInternal()->getBytesPerFrame();
mDistributionBuffer = new uint8_t[distributionBufferSizeBytes];
@@ -69,7 +69,7 @@
// Read audio data from stream using a blocking read.
result = getStreamInternal()->read(mDistributionBuffer, getFramesPerBurst(), timeoutNanos);
if (result == AAUDIO_ERROR_DISCONNECTED) {
- disconnectRegisteredStreams();
+ ALOGV("%s() read() returned AAUDIO_ERROR_DISCONNECTED, break", __func__);
break;
} else if (result != getFramesPerBurst()) {
ALOGW("callbackLoop() read %d / %d",
diff --git a/services/oboeservice/AAudioServiceEndpointCapture.h b/services/oboeservice/AAudioServiceEndpointCapture.h
index 971da9a..2bbe81d 100644
--- a/services/oboeservice/AAudioServiceEndpointCapture.h
+++ b/services/oboeservice/AAudioServiceEndpointCapture.h
@@ -36,8 +36,7 @@
void *callbackLoop() override;
private:
- AudioStreamInternalCapture mStreamInternalCapture;
- uint8_t *mDistributionBuffer = nullptr;
+ uint8_t *mDistributionBuffer = nullptr;
};
} /* namespace aaudio */
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 0843e0b..04c6453 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -226,7 +226,7 @@
return result;
}
-aaudio_result_t AAudioServiceEndpointMMAP::close() {
+void AAudioServiceEndpointMMAP::close() {
if (mMmapStream != nullptr) {
// Needs to be explicitly cleared or CTS will fail but it is not clear why.
mMmapStream.clear();
@@ -235,8 +235,6 @@
// FIXME Make closing synchronous.
AudioClock::sleepForNanos(100 * AAUDIO_NANOS_PER_MILLISECOND);
}
-
- return AAUDIO_OK;
}
aaudio_result_t AAudioServiceEndpointMMAP::startStream(sp<AAudioServiceStreamBase> stream,
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.h b/services/oboeservice/AAudioServiceEndpointMMAP.h
index 3d10861..b6003b6 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.h
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.h
@@ -50,7 +50,7 @@
aaudio_result_t open(const aaudio::AAudioStreamRequest &request) override;
- aaudio_result_t close() override;
+ void close() override;
aaudio_result_t startStream(android::sp<AAudioServiceStreamBase> stream,
audio_port_handle_t *clientHandle) override;
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
index bda4b90..dfe7193 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.cpp
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -42,8 +42,8 @@
#define BURSTS_PER_BUFFER_DEFAULT 2
AAudioServiceEndpointPlay::AAudioServiceEndpointPlay(AAudioService &audioService)
- : mStreamInternalPlay(audioService, true) {
- mStreamInternal = &mStreamInternalPlay;
+ : AAudioServiceEndpointShared(
+ (AudioStreamInternal *)(new AudioStreamInternalPlay(audioService, true))) {
}
aaudio_result_t AAudioServiceEndpointPlay::open(const aaudio::AAudioStreamRequest &request) {
@@ -145,7 +145,7 @@
result = getStreamInternal()->write(mMixer.getOutputBuffer(),
getFramesPerBurst(), timeoutNanos);
if (result == AAUDIO_ERROR_DISCONNECTED) {
- AAudioServiceEndpointShared::disconnectRegisteredStreams();
+ ALOGV("%s() write() returned AAUDIO_ERROR_DISCONNECTED, break", __func__);
break;
} else if (result != getFramesPerBurst()) {
ALOGW("callbackLoop() wrote %d / %d",
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.h b/services/oboeservice/AAudioServiceEndpointPlay.h
index 981e430..160a1de 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.h
+++ b/services/oboeservice/AAudioServiceEndpointPlay.h
@@ -45,7 +45,6 @@
void *callbackLoop() override;
private:
- AudioStreamInternalPlay mStreamInternalPlay; // for playing output of mixer
bool mLatencyTuningEnabled = false; // TODO implement tuning
AAudioMixer mMixer; //
};
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index dc21886..f5de59f 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -40,6 +40,9 @@
// This is the maximum size in frames. The effective size can be tuned smaller at runtime.
#define DEFAULT_BUFFER_CAPACITY (48 * 8)
+AAudioServiceEndpointShared::AAudioServiceEndpointShared(AudioStreamInternal *streamInternal)
+ : mStreamInternal(streamInternal) {}
+
std::string AAudioServiceEndpointShared::dump() const {
std::stringstream result;
@@ -84,8 +87,8 @@
return result;
}
-aaudio_result_t AAudioServiceEndpointShared::close() {
- return getStreamInternal()->releaseCloseFinal();
+void AAudioServiceEndpointShared::close() {
+ getStreamInternal()->releaseCloseFinal();
}
// Glue between C and C++ callbacks.
diff --git a/services/oboeservice/AAudioServiceEndpointShared.h b/services/oboeservice/AAudioServiceEndpointShared.h
index bfc1744..020b926 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.h
+++ b/services/oboeservice/AAudioServiceEndpointShared.h
@@ -35,12 +35,13 @@
class AAudioServiceEndpointShared : public AAudioServiceEndpoint {
public:
+ explicit AAudioServiceEndpointShared(AudioStreamInternal *streamInternal);
std::string dump() const override;
aaudio_result_t open(const aaudio::AAudioStreamRequest &request) override;
- aaudio_result_t close() override;
+ void close() override;
aaudio_result_t startStream(android::sp<AAudioServiceStreamBase> stream,
audio_port_handle_t *clientHandle) override;
@@ -57,15 +58,15 @@
protected:
AudioStreamInternal *getStreamInternal() const {
- return mStreamInternal;
+ return mStreamInternal.get();
};
aaudio_result_t startSharingThread_l();
aaudio_result_t stopSharingThread();
- // pointer to object statically allocated in subclasses
- AudioStreamInternal *mStreamInternal = nullptr;
+ // An MMAP stream that is shared by multiple clients.
+ android::sp<AudioStreamInternal> mStreamInternal;
std::atomic<bool> mCallbackEnabled{false};