Merge "Remove checks for specific sample rates and channel counts"
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 7e9d557..7d23d02 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -746,6 +746,7 @@
bool mInUnderrun; // whether track is currently in underrun state
String8 mName; // server's name for this IAudioTrack
+ uint32_t mPausedPosition;
private:
class DeathNotifier : public IBinder::DeathRecipient {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index d25c40b..3217171 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -99,7 +99,8 @@
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT)
+ mPreviousSchedulingGroup(SP_DEFAULT),
+ mPausedPosition(0)
{
}
@@ -121,7 +122,8 @@
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT)
+ mPreviousSchedulingGroup(SP_DEFAULT),
+ mPausedPosition(0)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
@@ -147,7 +149,8 @@
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT)
+ mPreviousSchedulingGroup(SP_DEFAULT),
+ mPausedPosition(0)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
@@ -551,6 +554,16 @@
}
mProxy->interrupt();
mAudioTrack->pause();
+
+ if (isOffloaded()) {
+ if (mOutput != 0) {
+ uint32_t halFrames;
+ // OffloadThread sends HAL pause in its threadLoop.. time saved
+ // here can be slightly off
+ AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
+ ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition);
+ }
+ }
}
status_t AudioTrack::setVolume(float left, float right)
@@ -770,6 +783,12 @@
if (isOffloaded_l()) {
uint32_t dspFrames = 0;
+ if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) {
+ ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition);
+ *position = mPausedPosition;
+ return NO_ERROR;
+ }
+
if (mOutput != 0) {
uint32_t halFrames;
AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
@@ -1488,6 +1507,7 @@
}
size_t misalignment = mProxy->getMisalignment();
uint32_t sequence = mSequence;
+ sp<AudioTrackClientProxy> proxy = mProxy;
// These fields don't need to be cached, because they are assigned only by set():
// mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags
@@ -1496,35 +1516,32 @@
mLock.unlock();
if (waitStreamEnd) {
- AutoMutex lock(mLock);
-
- sp<AudioTrackClientProxy> proxy = mProxy;
- sp<IMemory> iMem = mCblkMemory;
-
struct timespec timeout;
timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
timeout.tv_nsec = 0;
- mLock.unlock();
- status_t status = mProxy->waitStreamEndDone(&timeout);
- mLock.lock();
+ status_t status = proxy->waitStreamEndDone(&timeout);
switch (status) {
case NO_ERROR:
case DEAD_OBJECT:
case TIMED_OUT:
- mLock.unlock();
mCbf(EVENT_STREAM_END, mUserData, NULL);
- mLock.lock();
- if (mState == STATE_STOPPING) {
- mState = STATE_STOPPED;
- if (status != DEAD_OBJECT) {
- return NS_INACTIVE;
+ {
+ AutoMutex lock(mLock);
+ // The previously assigned value of waitStreamEnd is no longer valid,
+ // since the mutex has been unlocked and either the callback handler
+ // or another thread could have re-started the AudioTrack during that time.
+ waitStreamEnd = mState == STATE_STOPPING;
+ if (waitStreamEnd) {
+ mState = STATE_STOPPED;
}
}
- return 0;
- default:
- return 0;
+ if (waitStreamEnd && status != DEAD_OBJECT) {
+ return NS_INACTIVE;
+ }
+ break;
}
+ return 0;
}
// perform callbacks while unlocked
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 4be3c09..1a027a6 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -476,10 +476,11 @@
case START_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
- uint32_t stream = data.readInt32();
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
int session = data.readInt32();
reply->writeInt32(static_cast <uint32_t>(startOutput(output,
- (audio_stream_type_t)stream,
+ stream,
session)));
return NO_ERROR;
} break;
@@ -487,10 +488,11 @@
case STOP_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
- uint32_t stream = data.readInt32();
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
int session = data.readInt32();
reply->writeInt32(static_cast <uint32_t>(stopOutput(output,
- (audio_stream_type_t)stream,
+ stream,
session)));
return NO_ERROR;
} break;
@@ -633,7 +635,7 @@
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
uint32_t inPastMs = (uint32_t)data.readInt32();
- reply->writeInt32( isStreamActive((audio_stream_type_t) stream, inPastMs) );
+ reply->writeInt32( isStreamActive(stream, inPastMs) );
return NO_ERROR;
} break;
@@ -641,7 +643,7 @@
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
uint32_t inPastMs = (uint32_t)data.readInt32();
- reply->writeInt32( isStreamActiveRemotely((audio_stream_type_t) stream, inPastMs) );
+ reply->writeInt32( isStreamActiveRemotely(stream, inPastMs) );
return NO_ERROR;
} break;
diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index aeecdbc..a3093d0 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -35,7 +35,6 @@
#include <gui/SurfaceComposerClient.h>
#include <binder/ProcessState.h>
-#include <ui/FramebufferNativeWindow.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 357ea22..690d0d6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5555,12 +5555,12 @@
mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mBufferSize / mFrameSize;
// This is the formula for calculating the temporary buffer size.
- // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to
+ // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
// 1 full output buffer, regardless of the alignment of the available input.
- // The "3" is somewhat arbitrary, and could probably be larger.
+ // The value is somewhat arbitrary, and could probably be even larger.
// A larger value should allow more old data to be read after a track calls start(),
// without increasing latency.
- mRsmpInFrames = mFrameCount * 3;
+ mRsmpInFrames = mFrameCount * 7;
mRsmpInFramesP2 = roundup(mRsmpInFrames);
delete[] mRsmpInBuffer;
// Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer