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-rw-r--r--media/libmedia/AudioTrack.cpp291
1 files changed, 208 insertions, 83 deletions
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index cd7bcd5..587c8ff 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -41,6 +41,35 @@
#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
namespace android {
+// ---------------------------------------------------------------------------
+
+// static
+status_t AudioTrack::getMinFrameCount(
+ int* frameCount,
+ int streamType,
+ uint32_t sampleRate)
+{
+ 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;
+ }
+
+ // Ensure that buffer depth covers at least audio hardware latency
+ uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
+ if (minBufCount < 2) minBufCount = 2;
+
+ *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
+ afFrameCount * minBufCount * sampleRate / afSampleRate;
+ return NO_ERROR;
+}
// ---------------------------------------------------------------------------
@@ -58,11 +87,13 @@ AudioTrack::AudioTrack(
uint32_t flags,
callback_t cbf,
void* user,
- int notificationFrames)
+ int notificationFrames,
+ int sessionId)
: mStatus(NO_INIT)
{
mStatus = set(streamType, sampleRate, format, channels,
- frameCount, flags, cbf, user, notificationFrames, 0);
+ frameCount, flags, cbf, user, notificationFrames,
+ 0, false, sessionId);
}
AudioTrack::AudioTrack(
@@ -74,11 +105,13 @@ AudioTrack::AudioTrack(
uint32_t flags,
callback_t cbf,
void* user,
- int notificationFrames)
+ int notificationFrames,
+ int sessionId)
: mStatus(NO_INIT)
{
mStatus = set(streamType, sampleRate, format, channels,
- 0, flags, cbf, user, notificationFrames, sharedBuffer);
+ 0, flags, cbf, user, notificationFrames,
+ sharedBuffer, false, sessionId);
}
AudioTrack::~AudioTrack()
@@ -110,7 +143,8 @@ status_t AudioTrack::set(
void* user,
int notificationFrames,
const sp<IMemory>& sharedBuffer,
- bool threadCanCallJava)
+ bool threadCanCallJava,
+ int sessionId)
{
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
@@ -124,10 +158,6 @@ status_t AudioTrack::set(
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 +203,17 @@ status_t AudioTrack::set(
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;
+ mSendLevel = 0;
+ mFrameCount = frameCount;
+ mNotificationFramesReq = notificationFrames;
+ mSessionId = sessionId;
+ mAuxEffectId = 0;
+
// 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 +237,7 @@ status_t AudioTrack::set(
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 +277,7 @@ int AudioTrack::channelCount() const
uint32_t AudioTrack::frameCount() const
{
- return mFrameCount;
+ return mCblk->frameCount;
}
int AudioTrack::frameSize() const
@@ -303,6 +299,7 @@ sp<IMemory>& AudioTrack::sharedBuffer()
void AudioTrack::start()
{
sp<AudioTrackThread> t = mAudioTrackThread;
+ status_t status;
LOGV("start %p", this);
if (t != 0) {
@@ -319,17 +316,25 @@ void AudioTrack::start()
mNewPosition = mCblk->server + mUpdatePeriod;
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
+ mCblk->flags &= ~CBLK_DISABLED_ON;
if (t != 0) {
t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
} else {
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) {
@@ -430,19 +435,50 @@ bool AudioTrack::muted() const
return mMuted;
}
-void AudioTrack::setVolume(float left, float right)
+status_t AudioTrack::setVolume(float left, float right)
{
+ if (left > 1.0f || right > 1.0f) {
+ return BAD_VALUE;
+ }
+
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
// write must be atomic
- mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000);
+ mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
+
+ return NO_ERROR;
}
void AudioTrack::getVolume(float* left, float* right)
{
- *left = mVolume[LEFT];
- *right = mVolume[RIGHT];
+ if (left != NULL) {
+ *left = mVolume[LEFT];
+ }
+ if (right != NULL) {
+ *right = mVolume[RIGHT];
+ }
+}
+
+status_t AudioTrack::setAuxEffectSendLevel(float level)
+{
+ LOGV("setAuxEffectSendLevel(%f)", level);
+ if (level > 1.0f) {
+ return BAD_VALUE;
+ }
+
+ mSendLevel = level;
+
+ mCblk->sendLevel = uint16_t(level * 0x1000);
+
+ return NO_ERROR;
+}
+
+void AudioTrack::getAuxEffectSendLevel(float* level)
+{
+ if (level != NULL) {
+ *level = mSendLevel;
+ }
}
status_t AudioTrack::setSampleRate(int rate)
@@ -479,14 +515,14 @@ status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount
}
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 +602,7 @@ status_t AudioTrack::setPosition(uint32_t position)
if (position > mCblk->user) return BAD_VALUE;
mCblk->server = position;
- mCblk->forceReady = 1;
+ mCblk->flags |= CBLK_FORCEREADY_ON;
return NO_ERROR;
}
@@ -586,7 +622,7 @@ status_t AudioTrack::reload()
flush();
- mCblk->stepUser(mFrameCount);
+ mCblk->stepUser(mCblk->frameCount);
return NO_ERROR;
}
@@ -597,6 +633,21 @@ audio_io_handle_t AudioTrack::getOutput()
mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
}
+int AudioTrack::getSessionId()
+{
+ return mSessionId;
+}
+
+status_t AudioTrack::attachAuxEffect(int effectId)
+{
+ LOGV("attachAuxEffect(%d)", effectId);
+ status_t status = mAudioTrack->attachAuxEffect(effectId);
+ if (status == NO_ERROR) {
+ mAuxEffectId = effectId;
+ }
+ return status;
+}
+
// -------------------------------------------------------------------------
status_t AudioTrack::createTrack(
@@ -607,7 +658,8 @@ status_t AudioTrack::createTrack(
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 +668,61 @@ status_t AudioTrack::createTrack(
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,
@@ -625,6 +732,7 @@ status_t AudioTrack::createTrack(
((uint16_t)flags) << 16,
sharedBuffer,
output,
+ &mSessionId,
&status);
if (track == 0) {
@@ -641,20 +749,22 @@ status_t AudioTrack::createTrack(
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->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
+ mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
+ mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
+ mRemainingFrames = mNotificationFramesAct;
+ mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
return NO_ERROR;
}
@@ -685,8 +795,15 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
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 +817,9 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
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;
@@ -725,6 +843,13 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
cblk->lock.unlock();
}
+ // restart track if it was disabled by audioflinger due to previous underrun
+ if (cblk->flags & CBLK_DISABLED_MSK) {
+ cblk->flags &= ~CBLK_DISABLED_ON;
+ LOGW("obtainBuffer() track %p disabled, restarting", this);
+ mAudioTrack->start();
+ }
+
cblk->waitTimeMs = 0;
if (framesReq > framesAvail) {
@@ -826,13 +951,13 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
// 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 +1057,7 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
while (frames);
if (frames == 0) {
- mRemainingFrames = mNotificationFrames;
+ mRemainingFrames = mNotificationFramesAct;
} else {
mRemainingFrames = frames;
}
@@ -949,7 +1074,7 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
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 +1111,7 @@ audio_track_cblk_t::audio_track_cblk_t()
: 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), sendLevel(0)
{
}
@@ -996,7 +1121,7 @@ uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
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 +1138,7 @@ uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
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 +1163,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
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 +1214,7 @@ uint32_t audio_track_cblk_t::framesAvailable_l()
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 +1227,7 @@ uint32_t audio_track_cblk_t::framesReady()
uint32_t u = this->user;
uint32_t s = this->server;
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
return u - s;
} else {
generated by cgit v1.1 at 2025-08-04 08:54:52 +0000