Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock.
The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).
The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.
AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.
AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.
AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.
Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index cd7bcd5..c350532 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -124,10 +124,6 @@
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
- int afFrameCount;
- if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
- return NO_INIT;
- }
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
@@ -173,48 +169,13 @@
return BAD_VALUE;
}
- if (!AudioSystem::isLinearPCM(format)) {
- if (sharedBuffer != 0) {
- frameCount = sharedBuffer->size();
- }
- } else {
- // Ensure that buffer depth covers at least audio hardware latency
- uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
- if (minBufCount < 2) minBufCount = 2;
-
- int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
-
- if (sharedBuffer == 0) {
- if (frameCount == 0) {
- frameCount = minFrameCount;
- }
- if (notificationFrames == 0) {
- notificationFrames = frameCount/2;
- }
- // Make sure that application is notified with sufficient margin
- // before underrun
- if (notificationFrames > frameCount/2) {
- notificationFrames = frameCount/2;
- }
- if (frameCount < minFrameCount) {
- LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
- return BAD_VALUE;
- }
- } else {
- // Ensure that buffer alignment matches channelcount
- if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
- LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
- return BAD_VALUE;
- }
- frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
- }
- }
-
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
+ mFrameCount = frameCount;
+ mNotificationFramesReq = notificationFrames;
// create the IAudioTrack
status_t status = createTrack(streamType, sampleRate, format, channelCount,
- frameCount, flags, sharedBuffer, output);
+ frameCount, flags, sharedBuffer, output, true);
if (status != NO_ERROR) {
return status;
@@ -238,10 +199,7 @@
mMuted = false;
mActive = 0;
mCbf = cbf;
- mNotificationFrames = notificationFrames;
- mRemainingFrames = notificationFrames;
mUserData = user;
- mLatency = afLatency + (1000*mFrameCount) / sampleRate;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
@@ -281,7 +239,7 @@
uint32_t AudioTrack::frameCount() const
{
- return mFrameCount;
+ return mCblk->frameCount;
}
int AudioTrack::frameSize() const
@@ -303,6 +261,7 @@
void AudioTrack::start()
{
sp<AudioTrackThread> t = mAudioTrackThread;
+ status_t status;
LOGV("start %p", this);
if (t != 0) {
@@ -325,11 +284,18 @@
setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
}
- status_t status = mAudioTrack->start();
+ if (mCblk->flags & CBLK_INVALID_MSK) {
+ LOGW("start() track %p invalidated, creating a new one", this);
+ // no need to clear the invalid flag as this cblk will not be used anymore
+ // force new track creation
+ status = DEAD_OBJECT;
+ } else {
+ status = mAudioTrack->start();
+ }
if (status == DEAD_OBJECT) {
LOGV("start() dead IAudioTrack: creating a new one");
status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
- mFrameCount, mFlags, mSharedBuffer, getOutput());
+ mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
if (status == NO_ERROR) {
status = mAudioTrack->start();
if (status == NO_ERROR) {
@@ -479,14 +445,14 @@
}
if (loopStart >= loopEnd ||
- loopEnd - loopStart > mFrameCount) {
- LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
+ loopEnd - loopStart > cblk->frameCount) {
+ LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
return BAD_VALUE;
}
- if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) {
+ if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
- loopStart, loopEnd, mFrameCount);
+ loopStart, loopEnd, cblk->frameCount);
return BAD_VALUE;
}
@@ -566,7 +532,7 @@
if (position > mCblk->user) return BAD_VALUE;
mCblk->server = position;
- mCblk->forceReady = 1;
+ mCblk->flags |= CBLK_FORCEREADY_ON;
return NO_ERROR;
}
@@ -586,7 +552,7 @@
flush();
- mCblk->stepUser(mFrameCount);
+ mCblk->stepUser(mCblk->frameCount);
return NO_ERROR;
}
@@ -607,7 +573,8 @@
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output)
+ audio_io_handle_t output,
+ bool enforceFrameCount)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -616,6 +583,61 @@
return NO_INIT;
}
+ int afSampleRate;
+ if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ int afFrameCount;
+ if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ uint32_t afLatency;
+ if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+
+ mNotificationFramesAct = mNotificationFramesReq;
+ if (!AudioSystem::isLinearPCM(format)) {
+ if (sharedBuffer != 0) {
+ frameCount = sharedBuffer->size();
+ }
+ } else {
+ // Ensure that buffer depth covers at least audio hardware latency
+ uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
+ if (minBufCount < 2) minBufCount = 2;
+
+ int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+
+ if (sharedBuffer == 0) {
+ if (frameCount == 0) {
+ frameCount = minFrameCount;
+ }
+ if (mNotificationFramesAct == 0) {
+ mNotificationFramesAct = frameCount/2;
+ }
+ // Make sure that application is notified with sufficient margin
+ // before underrun
+ if (mNotificationFramesAct > (uint32_t)frameCount/2) {
+ mNotificationFramesAct = frameCount/2;
+ }
+ if (frameCount < minFrameCount) {
+ if (enforceFrameCount) {
+ LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
+ return BAD_VALUE;
+ } else {
+ frameCount = minFrameCount;
+ }
+ }
+ } else {
+ // Ensure that buffer alignment matches channelcount
+ if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
+ LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
+ return BAD_VALUE;
+ }
+ frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
+ }
+ }
+
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
@@ -641,20 +663,20 @@
mCblkMemory.clear();
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
- mCblk->out = 1;
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount;
+ mCblk->flags |= CBLK_DIRECTION_OUT;
if (sharedBuffer == 0) {
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
} else {
mCblk->buffers = sharedBuffer->pointer();
// Force buffer full condition as data is already present in shared memory
- mCblk->stepUser(mFrameCount);
+ mCblk->stepUser(mCblk->frameCount);
}
mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
+ mRemainingFrames = mNotificationFramesAct;
+ mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
return NO_ERROR;
}
@@ -685,8 +707,15 @@
cblk->lock.unlock();
return WOULD_BLOCK;
}
-
- result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+ if (!(cblk->flags & CBLK_INVALID_MSK)) {
+ result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+ }
+ if (cblk->flags & CBLK_INVALID_MSK) {
+ LOGW("obtainBuffer() track %p invalidated, creating a new one", this);
+ // no need to clear the invalid flag as this cblk will not be used anymore
+ cblk->lock.unlock();
+ goto create_new_track;
+ }
if (__builtin_expect(result!=NO_ERROR, false)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
@@ -700,8 +729,9 @@
result = mAudioTrack->start();
if (result == DEAD_OBJECT) {
LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
+create_new_track:
result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
- mFrameCount, mFlags, mSharedBuffer, getOutput());
+ mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
if (result == NO_ERROR) {
cblk = mCblk;
cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
@@ -826,13 +856,13 @@
// Manage underrun callback
if (mActive && (mCblk->framesReady() == 0)) {
- LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
- if (mCblk->flowControlFlag == 0) {
+ LOGV("Underrun user: %x, server: %x, flags %04x", mCblk->user, mCblk->server, mCblk->flags);
+ if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
if (mCblk->server == mCblk->frameCount) {
mCbf(EVENT_BUFFER_END, mUserData, 0);
}
- mCblk->flowControlFlag = 1;
+ mCblk->flags |= CBLK_UNDERRUN_ON;
if (mSharedBuffer != 0) return false;
}
}
@@ -932,7 +962,7 @@
while (frames);
if (frames == 0) {
- mRemainingFrames = mNotificationFrames;
+ mRemainingFrames = mNotificationFramesAct;
} else {
mRemainingFrames = frames;
}
@@ -949,7 +979,7 @@
result.append(" AudioTrack::dump\n");
snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
result.append(buffer);
- snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
+ snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
result.append(buffer);
@@ -986,7 +1016,7 @@
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
userBase(0), serverBase(0), buffers(0), frameCount(0),
loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
- flowControlFlag(1), forceReady(0)
+ flags(0)
{
}
@@ -996,7 +1026,7 @@
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
// If stepServer() has been called once, switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
@@ -1013,7 +1043,7 @@
this->user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
- flowControlFlag = 0;
+ flags &= ~CBLK_UNDERRUN_MSK;
return u;
}
@@ -1038,7 +1068,7 @@
uint32_t s = this->server;
s += frameCount;
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
// Mark that we have read the first buffer so that next time stepUser() is called
// we switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
@@ -1089,7 +1119,7 @@
uint32_t u = this->user;
uint32_t s = this->server;
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
return limit + frameCount - u;
} else {
@@ -1102,7 +1132,7 @@
uint32_t u = this->user;
uint32_t s = this->server;
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
return u - s;
} else {