Remove TimedAudioTrack and associated code
Bug: 8278435
Change-Id: I095c1a4888e645e14d93b0b15fbef4524a831ca1
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 87f9aaa..6e3eb83 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -41,7 +41,6 @@
libaudioresampler \
libaudiospdif \
libaudioutils \
- libcommon_time_client \
libcutils \
libutils \
liblog \
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d2786b9..f4bd1c4 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -56,8 +56,6 @@
#include <powermanager/PowerManager.h>
-#include <common_time/cc_helper.h>
-
#include <media/IMediaLogService.h>
#include <media/nbaio/Pipe.h>
@@ -1359,8 +1357,7 @@
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
- mPid(pid),
- mTimedTrackCount(0)
+ mPid(pid)
{
size_t heapSize = kClientSharedHeapSizeBytes;
// Increase heap size on non low ram devices to limit risk of reconnection failure for
@@ -1382,31 +1379,6 @@
return mMemoryDealer;
}
-// Reserve one of the limited slots for a timed audio track associated
-// with this client
-bool AudioFlinger::Client::reserveTimedTrack()
-{
- const int kMaxTimedTracksPerClient = 4;
-
- Mutex::Autolock _l(mTimedTrackLock);
-
- if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
- ALOGW("can not create timed track - pid %d has exceeded the limit",
- mPid);
- return false;
- }
-
- mTimedTrackCount++;
- return true;
-}
-
-// Release a slot for a timed audio track
-void AudioFlinger::Client::releaseTimedTrack()
-{
- Mutex::Autolock _l(mTimedTrackLock);
- mTimedTrackCount--;
-}
-
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
@@ -2979,8 +2951,7 @@
void *buffer = malloc(TEE_SINK_READ * frameSize);
for (;;) {
size_t count = TEE_SINK_READ;
- ssize_t actual = teeSource->read(buffer, count,
- AudioBufferProvider::kInvalidPTS);
+ ssize_t actual = teeSource->read(buffer, count);
bool wasFirstRead = firstRead;
firstRead = false;
if (actual <= 0) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 2571e67..62a3115 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -23,8 +23,6 @@
#include <sys/types.h>
#include <limits.h>
-#include <common_time/cc_helper.h>
-
#include <cutils/compiler.h>
#include <media/IAudioFlinger.h>
@@ -414,18 +412,12 @@
pid_t pid() const { return mPid; }
sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
- bool reserveTimedTrack();
- void releaseTimedTrack();
-
private:
Client(const Client&);
Client& operator = (const Client&);
const sp<AudioFlinger> mAudioFlinger;
sp<MemoryDealer> mMemoryDealer;
const pid_t mPid;
-
- Mutex mTimedTrackLock;
- int mTimedTrackCount;
};
// --- Notification Client ---
@@ -496,12 +488,6 @@
virtual void flush();
virtual void pause();
virtual status_t attachAuxEffect(int effectId);
- virtual status_t allocateTimedBuffer(size_t size,
- sp<IMemory>* buffer);
- virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
- int64_t pts);
- virtual status_t setMediaTimeTransform(const LinearTransform& xform,
- int target);
virtual status_t setParameters(const String8& keyValuePairs);
virtual status_t getTimestamp(AudioTimestamp& timestamp);
virtual void signal(); // signal playback thread for a change in control block
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 8a9a837..aea6b67 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -36,8 +36,6 @@
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
-#include <common_time/local_clock.h>
-#include <common_time/cc_helper.h>
#include "AudioMixerOps.h"
#include "AudioMixer.h"
@@ -786,7 +784,6 @@
mMixerInFormat,
resamplerChannelCount,
devSampleRate, quality);
- resampler->setLocalTimeFreq(sLocalTimeFreq);
}
return true;
}
@@ -906,13 +903,13 @@
}
-void AudioMixer::process(int64_t pts)
+void AudioMixer::process()
{
- mState.hook(&mState, pts);
+ mState.hook(&mState);
}
-void AudioMixer::process__validate(state_t* state, int64_t pts)
+void AudioMixer::process__validate(state_t* state)
{
ALOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
@@ -1042,7 +1039,7 @@
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state, pts);
+ state->hook(state);
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
@@ -1367,7 +1364,7 @@
}
// no-op case
-void AudioMixer::process__nop(state_t* state, int64_t pts)
+void AudioMixer::process__nop(state_t* state)
{
ALOGVV("process__nop\n");
uint32_t e0 = state->enabledTracks;
@@ -1401,9 +1398,7 @@
size_t outFrames = state->frameCount;
while (outFrames) {
t3.buffer.frameCount = outFrames;
- int64_t outputPTS = calculateOutputPTS(
- t3, pts, state->frameCount - outFrames);
- t3.bufferProvider->getNextBuffer(&t3.buffer, outputPTS);
+ t3.bufferProvider->getNextBuffer(&t3.buffer);
if (t3.buffer.raw == NULL) break;
outFrames -= t3.buffer.frameCount;
t3.bufferProvider->releaseBuffer(&t3.buffer);
@@ -1414,7 +1409,7 @@
}
// generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
+void AudioMixer::process__genericNoResampling(state_t* state)
{
ALOGVV("process__genericNoResampling\n");
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
@@ -1427,7 +1422,7 @@
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
- t.bufferProvider->getNextBuffer(&t.buffer, pts);
+ t.bufferProvider->getNextBuffer(&t.buffer);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
}
@@ -1486,9 +1481,7 @@
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) -
(BLOCKSIZE - outFrames);
- int64_t outputPTS = calculateOutputPTS(
- t, pts, numFrames + (BLOCKSIZE - outFrames));
- t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
+ t.bufferProvider->getNextBuffer(&t.buffer);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
@@ -1522,7 +1515,7 @@
// generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
+void AudioMixer::process__genericResampling(state_t* state)
{
ALOGVV("process__genericResampling\n");
// this const just means that local variable outTemp doesn't change
@@ -1561,7 +1554,6 @@
// acquire/release the buffers because it's done by
// the resampler.
if (t.needs & NEEDS_RESAMPLE) {
- t.resampler->setPTS(pts);
t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
@@ -1569,8 +1561,7 @@
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
- int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
- t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
+ t.bufferProvider->getNextBuffer(&t.buffer);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
@@ -1592,8 +1583,7 @@
}
// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
- int64_t pts)
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
{
ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
// This method is only called when state->enabledTracks has exactly
@@ -1615,8 +1605,7 @@
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
- int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
- t.bufferProvider->getNextBuffer(&b, outputPTS);
+ t.bufferProvider->getNextBuffer(&b);
const int16_t *in = b.i16;
// in == NULL can happen if the track was flushed just after having
@@ -1677,24 +1666,10 @@
}
}
-int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
- int outputFrameIndex)
-{
- if (AudioBufferProvider::kInvalidPTS == basePTS) {
- return AudioBufferProvider::kInvalidPTS;
- }
-
- return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
-}
-
-/*static*/ uint64_t AudioMixer::sLocalTimeFreq;
/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
/*static*/ void AudioMixer::sInitRoutine()
{
- LocalClock lc;
- sLocalTimeFreq = lc.getLocalFreq(); // for the resampler
-
DownmixerBufferProvider::init(); // for the downmixer
}
@@ -1836,7 +1811,7 @@
* TA: int32_t (Q4.27)
*/
template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::process_NoResampleOneTrack(state_t* state, int64_t pts)
+void AudioMixer::process_NoResampleOneTrack(state_t* state)
{
ALOGVV("process_NoResampleOneTrack\n");
// CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
@@ -1852,8 +1827,7 @@
AudioBufferProvider::Buffer& b(t->buffer);
// get input buffer
b.frameCount = numFrames;
- const int64_t outputPTS = calculateOutputPTS(*t, pts, state->frameCount - numFrames);
- t->bufferProvider->getNextBuffer(&b, outputPTS);
+ t->bufferProvider->getNextBuffer(&b);
const TI *in = reinterpret_cast<TI*>(b.raw);
// in == NULL can happen if the track was flushed just after having
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 7165c6c..e788ac3 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -126,7 +126,7 @@
void setParameter(int name, int target, int param, void *value);
void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
- void process(int64_t pts);
+ void process();
uint32_t trackNames() const { return mTrackNames; }
@@ -278,7 +278,7 @@
void reconfigureBufferProviders();
};
- typedef void (*process_hook_t)(state_t* state, int64_t pts);
+ typedef void (*process_hook_t)(state_t* state);
// pad to 32-bytes to fill cache line
struct state_t {
@@ -328,17 +328,12 @@
static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux);
- static void process__validate(state_t* state, int64_t pts);
- static void process__nop(state_t* state, int64_t pts);
- static void process__genericNoResampling(state_t* state, int64_t pts);
- static void process__genericResampling(state_t* state, int64_t pts);
- static void process__OneTrack16BitsStereoNoResampling(state_t* state,
- int64_t pts);
+ static void process__validate(state_t* state);
+ static void process__nop(state_t* state);
+ static void process__genericNoResampling(state_t* state);
+ static void process__genericResampling(state_t* state);
+ static void process__OneTrack16BitsStereoNoResampling(state_t* state);
- static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
- int outputFrameIndex);
-
- static uint64_t sLocalTimeFreq;
static pthread_once_t sOnceControl;
static void sInitRoutine();
@@ -359,7 +354,7 @@
// multi-format process hooks
template <int MIXTYPE, typename TO, typename TI, typename TA>
- static void process_NoResampleOneTrack(state_t* state, int64_t pts);
+ static void process_NoResampleOneTrack(state_t* state);
// multi-format track hooks
template <int MIXTYPE, typename TO, typename TI, typename TA>
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index e49b7b1..7c20478 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -261,8 +261,8 @@
int32_t sampleRate, src_quality quality) :
mChannelCount(inChannelCount),
mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
- mPhaseFraction(0), mLocalTimeFreq(0),
- mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
+ mPhaseFraction(0),
+ mQuality(quality) {
const int maxChannels = quality < DYN_LOW_QUALITY ? 2 : 8;
if (inChannelCount < 1
@@ -304,23 +304,6 @@
mVolume[1] = u4_12_from_float(clampFloatVol(right));
}
-void AudioResampler::setLocalTimeFreq(uint64_t freq) {
- mLocalTimeFreq = freq;
-}
-
-void AudioResampler::setPTS(int64_t pts) {
- mPTS = pts;
-}
-
-int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
-
- if (mPTS == AudioBufferProvider::kInvalidPTS) {
- return AudioBufferProvider::kInvalidPTS;
- } else {
- return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
- }
-}
-
void AudioResampler::reset() {
mInputIndex = 0;
mPhaseFraction = 0;
@@ -368,8 +351,7 @@
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / 2));
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
goto resampleStereo16_exit;
}
@@ -465,8 +447,7 @@
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / 2));
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index a8e3e6f..c4627e8 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -59,10 +59,6 @@
virtual void init() = 0;
virtual void setSampleRate(int32_t inSampleRate);
virtual void setVolume(float left, float right);
- virtual void setLocalTimeFreq(uint64_t freq);
-
- // set the PTS of the next buffer output by the resampler
- virtual void setPTS(int64_t pts);
// Resample int16_t samples from provider and accumulate into 'out'.
// A mono provider delivers a sequence of samples.
@@ -103,8 +99,6 @@
AudioResampler(const AudioResampler&);
AudioResampler& operator=(const AudioResampler&);
- int64_t calculateOutputPTS(int outputFrameIndex);
-
const int32_t mChannelCount;
const int32_t mSampleRate;
int32_t mInSampleRate;
@@ -117,8 +111,6 @@
size_t mInputIndex;
int32_t mPhaseIncrement;
uint32_t mPhaseFraction;
- uint64_t mLocalTimeFreq;
- int64_t mPTS;
// returns the inFrameCount required to generate outFrameCount frames.
//
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 172c2a5..6a324ad 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -66,7 +66,7 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer, mPTS);
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
return 0;
}
@@ -97,8 +97,7 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / 2));
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
goto save_state; // ugly, but efficient
}
@@ -135,7 +134,7 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer, mPTS);
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
return 0;
}
@@ -166,8 +165,7 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / 2));
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
goto save_state; // ugly, but efficient
}
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 6481b85..618b56c 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -527,8 +527,7 @@
// We may not fetch a new buffer if the existing data is sufficient.
while (mBuffer.frameCount == 0 && inFrameCount > 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / OUTPUT_CHANNELS));
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
goto resample_exit;
}
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 41730ee..e93c064 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -301,8 +301,7 @@
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / 2));
+ provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL) {
goto resample_exit;
}
diff --git a/services/audioflinger/BufferProviders.cpp b/services/audioflinger/BufferProviders.cpp
index a8be206..2ca2cac 100644
--- a/services/audioflinger/BufferProviders.cpp
+++ b/services/audioflinger/BufferProviders.cpp
@@ -70,13 +70,12 @@
free(mLocalBufferData);
}
-status_t CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
- int64_t pts)
+status_t CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer)
{
- //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
- // this, pBuffer, pBuffer->frameCount, pts);
+ //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu))",
+ // this, pBuffer, pBuffer->frameCount);
if (mLocalBufferFrameCount == 0) {
- status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
+ status_t res = mTrackBufferProvider->getNextBuffer(pBuffer);
if (res == OK) {
copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
}
@@ -84,7 +83,7 @@
}
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = pBuffer->frameCount;
- status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
+ status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
// At one time an upstream buffer provider had
// res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
//
@@ -356,13 +355,13 @@
}
status_t TimestretchBufferProvider::getNextBuffer(
- AudioBufferProvider::Buffer *pBuffer, int64_t pts)
+ AudioBufferProvider::Buffer *pBuffer)
{
- ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
- this, pBuffer, pBuffer->frameCount, pts);
+ ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu))",
+ this, pBuffer, pBuffer->frameCount);
// BYPASS
- //return mTrackBufferProvider->getNextBuffer(pBuffer, pts);
+ //return mTrackBufferProvider->getNextBuffer(pBuffer);
// check if previously processed data is sufficient.
if (pBuffer->frameCount <= mRemaining) {
@@ -391,7 +390,7 @@
mBuffer.frameCount = mPlaybackRate.mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
? outputDesired : outputDesired * mPlaybackRate.mSpeed + 1;
- status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
+ status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
diff --git a/services/audioflinger/BufferProviders.h b/services/audioflinger/BufferProviders.h
index 4bc895c..abd43c6 100644
--- a/services/audioflinger/BufferProviders.h
+++ b/services/audioflinger/BufferProviders.h
@@ -64,7 +64,7 @@
virtual ~CopyBufferProvider();
// Overrides AudioBufferProvider methods
- virtual status_t getNextBuffer(Buffer *buffer, int64_t pts);
+ virtual status_t getNextBuffer(Buffer *buffer);
virtual void releaseBuffer(Buffer *buffer);
// Overrides PassthruBufferProvider
@@ -156,7 +156,7 @@
virtual ~TimestretchBufferProvider();
// Overrides AudioBufferProvider methods
- virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
+ virtual status_t getNextBuffer(Buffer* buffer);
virtual void releaseBuffer(Buffer* buffer);
// Overrides PassthruBufferProvider
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index 1bba5f6..bb83858 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -166,8 +166,7 @@
ALOG_ASSERT(mReadBuffer != NULL);
dumpState->mReadSequence++;
ATRACE_BEGIN("read");
- ssize_t framesRead = mInputSource->read(mReadBuffer, frameCount,
- AudioBufferProvider::kInvalidPTS);
+ ssize_t framesRead = mInputSource->read(mReadBuffer, frameCount);
ATRACE_END();
dumpState->mReadSequence++;
if (framesRead >= 0) {
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 3f99b43..cc9acff 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -402,13 +402,8 @@
ftDump->mFramesReady = framesReady;
}
- int64_t pts;
- if (mOutputSink == NULL || (OK != mOutputSink->getNextWriteTimestamp(&pts))) {
- pts = AudioBufferProvider::kInvalidPTS;
- }
-
// process() is CPU-bound
- mMixer->process(pts);
+ mMixer->process();
mMixerBufferState = MIXED;
} else if (mMixerBufferState == MIXED) {
mMixerBufferState = UNDEFINED;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 7bc6f0c..1450ca1 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -83,8 +83,7 @@
Track& operator = (const Track&);
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts = kInvalidPTS);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
// releaseBuffer() not overridden
// ExtendedAudioBufferProvider interface
@@ -158,92 +157,6 @@
}; // end of Track
-class TimedTrack : public Track {
- public:
- static sp<TimedTrack> create(PlaybackThread *thread,
- const sp<Client>& client,
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- const sp<IMemory>& sharedBuffer,
- int sessionId,
- int uid);
- virtual ~TimedTrack();
-
- class TimedBuffer {
- public:
- TimedBuffer();
- TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
- const sp<IMemory>& buffer() const { return mBuffer; }
- int64_t pts() const { return mPTS; }
- uint32_t position() const { return mPosition; }
- void setPosition(uint32_t pos) { mPosition = pos; }
- private:
- sp<IMemory> mBuffer;
- int64_t mPTS;
- uint32_t mPosition;
- };
-
- // Mixer facing methods.
- virtual size_t framesReady() const;
-
- // AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts);
- virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
-
- // Client/App facing methods.
- status_t allocateTimedBuffer(size_t size,
- sp<IMemory>* buffer);
- status_t queueTimedBuffer(const sp<IMemory>& buffer,
- int64_t pts);
- status_t setMediaTimeTransform(const LinearTransform& xform,
- TimedAudioTrack::TargetTimeline target);
-
- private:
- TimedTrack(PlaybackThread *thread,
- const sp<Client>& client,
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- const sp<IMemory>& sharedBuffer,
- int sessionId,
- int uid);
-
- void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
- void timedYieldSilence_l(uint32_t numFrames,
- AudioBufferProvider::Buffer* buffer);
- void trimTimedBufferQueue_l();
- void trimTimedBufferQueueHead_l(const char* logTag);
- void updateFramesPendingAfterTrim_l(const TimedBuffer& buf,
- const char* logTag);
-
- uint64_t mLocalTimeFreq;
- LinearTransform mLocalTimeToSampleTransform;
- LinearTransform mMediaTimeToSampleTransform;
- sp<MemoryDealer> mTimedMemoryDealer;
-
- Vector<TimedBuffer> mTimedBufferQueue;
- bool mQueueHeadInFlight;
- bool mTrimQueueHeadOnRelease;
- uint32_t mFramesPendingInQueue;
-
- uint8_t* mTimedSilenceBuffer;
- uint32_t mTimedSilenceBufferSize;
- mutable Mutex mTimedBufferQueueLock;
- bool mTimedAudioOutputOnTime;
- CCHelper mCCHelper;
-
- Mutex mMediaTimeTransformLock;
- LinearTransform mMediaTimeTransform;
- bool mMediaTimeTransformValid;
- TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
-};
-
// playback track, used by DuplicatingThread
class OutputTrack : public Track {
@@ -303,8 +216,7 @@
virtual ~PatchTrack();
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// PatchProxyBufferProvider interface
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 25d6d95..e2014b7 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -61,8 +61,7 @@
RecordTrack& operator = (const RecordTrack&);
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts = kInvalidPTS);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
// releaseBuffer() not overridden
bool mOverflow; // overflow on most recent attempt to fill client buffer
@@ -99,8 +98,7 @@
virtual ~PatchRecord();
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// PatchProxyBufferProvider interface
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 0458554..8ae798c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -53,9 +53,6 @@
#include <powermanager/PowerManager.h>
-#include <common_time/cc_helper.h>
-#include <common_time/local_clock.h>
-
#include "AudioFlinger.h"
#include "AudioMixer.h"
#include "BufferProviders.h"
@@ -1666,13 +1663,9 @@
sp<Track> track;
status_t lStatus;
- bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0;
-
// client expresses a preference for FAST, but we get the final say
if (*flags & IAudioFlinger::TRACK_FAST) {
if (
- // not timed
- (!isTimed) &&
// either of these use cases:
(
// use case 1: shared buffer with any frame count
@@ -1716,11 +1709,11 @@
ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
frameCount, mFrameCount);
} else {
- ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
+ ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%d "
"mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
"sampleRate=%u mSampleRate=%u "
"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
- isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
+ sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
audio_is_linear_pcm(format),
channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
*flags &= ~IAudioFlinger::TRACK_FAST;
@@ -1819,17 +1812,10 @@
}
}
- if (!isTimed) {
- track = new Track(this, client, streamType, sampleRate, format,
- channelMask, frameCount, NULL, sharedBuffer,
- sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);
- } else {
- track = TimedTrack::create(this, client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, sessionId, uid);
- }
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, NULL, sharedBuffer,
+ sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);
- // new Track always returns non-NULL,
- // but TimedTrack::create() is a factory that could fail by returning NULL
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
@@ -3640,22 +3626,8 @@
void AudioFlinger::MixerThread::threadLoop_mix()
{
- // obtain the presentation timestamp of the next output buffer
- int64_t pts;
- status_t status = INVALID_OPERATION;
-
- if (mNormalSink != 0) {
- status = mNormalSink->getNextWriteTimestamp(&pts);
- } else {
- status = mOutputSink->getNextWriteTimestamp(&pts);
- }
-
- if (status != NO_ERROR) {
- pts = AudioBufferProvider::kInvalidPTS;
- }
-
// mix buffers...
- mAudioMixer->process(pts);
+ mAudioMixer->process();
mCurrentWriteLength = mSinkBufferSize;
// increase sleep time progressively when application underrun condition clears.
// Only increase sleep time if the mixer is ready for two consecutive times to avoid
@@ -5369,7 +5341,7 @@
{
// mix buffers...
if (outputsReady(outputTracks)) {
- mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
+ mAudioMixer->process();
} else {
if (mMixerBufferValid) {
memset(mMixerBuffer, 0, mMixerBufferSize);
@@ -5909,7 +5881,7 @@
if (mPipeSource != 0) {
size_t framesToRead = mBufferSize / mFrameSize;
framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
- framesToRead, AudioBufferProvider::kInvalidPTS);
+ framesToRead);
if (framesRead == 0) {
// since pipe is non-blocking, simulate blocking input
sleepUs = (framesToRead * 1000000LL) / mSampleRate;
@@ -6531,7 +6503,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
+ AudioBufferProvider::Buffer* buffer)
{
sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
if (threadBase == 0) {
@@ -6632,7 +6604,7 @@
AudioBufferProvider::Buffer buffer;
for (size_t i = frames; i > 0; ) {
buffer.frameCount = i;
- status_t status = provider->getNextBuffer(&buffer, 0);
+ status_t status = provider->getNextBuffer(&buffer);
if (status != OK || buffer.frameCount == 0) {
frames -= i; // cannot fill request.
break;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 8eed50d..fa047fa 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1109,7 +1109,7 @@
virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL);
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
private:
RecordTrack * const mRecordTrack;
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 98bf96e..26067e3 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -50,7 +50,6 @@
enum track_type {
TYPE_DEFAULT,
- TYPE_TIMED,
TYPE_OUTPUT,
TYPE_PATCH,
};
@@ -83,7 +82,6 @@
sp<IMemory> getBuffers() const { return mBufferMemory; }
void* buffer() const { return mBuffer; }
bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; }
- bool isTimedTrack() const { return (mType == TYPE_TIMED); }
bool isOutputTrack() const { return (mType == TYPE_OUTPUT); }
bool isPatchTrack() const { return (mType == TYPE_PATCH); }
bool isExternalTrack() const { return !isOutputTrack() && !isPatchTrack(); }
@@ -93,7 +91,7 @@
TrackBase& operator = (const TrackBase&);
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0;
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// ExtendedAudioBufferProvider interface is only needed for Track,
@@ -132,7 +130,7 @@
}
bool isOut() const { return mIsOut; }
- // true for Track and TimedTrack, false for RecordTrack,
+ // true for Track, false for RecordTrack,
// this could be a track type if needed later
const wp<ThreadBase> mThread;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index c753afd..5830f75 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -27,9 +27,6 @@
#include <private/media/AudioTrackShared.h>
-#include <common_time/cc_helper.h>
-#include <common_time/local_clock.h>
-
#include "AudioMixer.h"
#include "AudioFlinger.h"
#include "ServiceUtilities.h"
@@ -242,7 +239,7 @@
// AudioBufferProvider interface
// getNextBuffer() = 0;
-// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
+// This implementation of releaseBuffer() is used by Track and RecordTrack
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
#ifdef TEE_SINK
@@ -308,43 +305,6 @@
return mTrack->attachAuxEffect(EffectId);
}
-status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
- sp<IMemory>* buffer) {
- if (!mTrack->isTimedTrack())
- return INVALID_OPERATION;
-
- PlaybackThread::TimedTrack* tt =
- reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
- return tt->allocateTimedBuffer(size, buffer);
-}
-
-status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
- int64_t pts) {
- if (!mTrack->isTimedTrack())
- return INVALID_OPERATION;
-
- if (buffer == 0 || buffer->pointer() == NULL) {
- ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
- return BAD_VALUE;
- }
-
- PlaybackThread::TimedTrack* tt =
- reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
- return tt->queueTimedBuffer(buffer, pts);
-}
-
-status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
- const LinearTransform& xform, int target) {
-
- if (!mTrack->isTimedTrack())
- return INVALID_OPERATION;
-
- PlaybackThread::TimedTrack* tt =
- reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
- return tt->setMediaTimeTransform(
- xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
-}
-
status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
return mTrack->setParameters(keyValuePairs);
}
@@ -590,7 +550,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
+ AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
size_t desiredFrames = buffer->frameCount;
@@ -1119,526 +1079,6 @@
}
// ----------------------------------------------------------------------------
-sp<AudioFlinger::PlaybackThread::TimedTrack>
-AudioFlinger::PlaybackThread::TimedTrack::create(
- PlaybackThread *thread,
- const sp<Client>& client,
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- const sp<IMemory>& sharedBuffer,
- int sessionId,
- int uid)
-{
- if (!client->reserveTimedTrack())
- return 0;
-
- return new TimedTrack(
- thread, client, streamType, sampleRate, format, channelMask, frameCount,
- sharedBuffer, sessionId, uid);
-}
-
-AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
- PlaybackThread *thread,
- const sp<Client>& client,
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- const sp<IMemory>& sharedBuffer,
- int sessionId,
- int uid)
- : Track(thread, client, streamType, sampleRate, format, channelMask,
- frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer,
- sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED),
- mQueueHeadInFlight(false),
- mTrimQueueHeadOnRelease(false),
- mFramesPendingInQueue(0),
- mTimedSilenceBuffer(NULL),
- mTimedSilenceBufferSize(0),
- mTimedAudioOutputOnTime(false),
- mMediaTimeTransformValid(false)
-{
- LocalClock lc;
- mLocalTimeFreq = lc.getLocalFreq();
-
- mLocalTimeToSampleTransform.a_zero = 0;
- mLocalTimeToSampleTransform.b_zero = 0;
- mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
- mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
- LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
- &mLocalTimeToSampleTransform.a_to_b_denom);
-
- mMediaTimeToSampleTransform.a_zero = 0;
- mMediaTimeToSampleTransform.b_zero = 0;
- mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
- mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
- LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
- &mMediaTimeToSampleTransform.a_to_b_denom);
-}
-
-AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
- mClient->releaseTimedTrack();
- delete [] mTimedSilenceBuffer;
-}
-
-status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
- size_t size, sp<IMemory>* buffer) {
-
- Mutex::Autolock _l(mTimedBufferQueueLock);
-
- trimTimedBufferQueue_l();
-
- // lazily initialize the shared memory heap for timed buffers
- if (mTimedMemoryDealer == NULL) {
- const int kTimedBufferHeapSize = 512 << 10;
-
- mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
- "AudioFlingerTimed");
- if (mTimedMemoryDealer == NULL) {
- return NO_MEMORY;
- }
- }
-
- sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
- if (newBuffer == 0 || newBuffer->pointer() == NULL) {
- return NO_MEMORY;
- }
-
- *buffer = newBuffer;
- return NO_ERROR;
-}
-
-// caller must hold mTimedBufferQueueLock
-void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
- int64_t mediaTimeNow;
- {
- Mutex::Autolock mttLock(mMediaTimeTransformLock);
- if (!mMediaTimeTransformValid)
- return;
-
- int64_t targetTimeNow;
- status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
- ? mCCHelper.getCommonTime(&targetTimeNow)
- : mCCHelper.getLocalTime(&targetTimeNow);
-
- if (OK != res)
- return;
-
- if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
- &mediaTimeNow)) {
- return;
- }
- }
-
- size_t trimEnd;
- for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
- int64_t bufEnd;
-
- if ((trimEnd + 1) < mTimedBufferQueue.size()) {
- // We have a next buffer. Just use its PTS as the PTS of the frame
- // following the last frame in this buffer. If the stream is sparse
- // (ie, there are deliberate gaps left in the stream which should be
- // filled with silence by the TimedAudioTrack), then this can result
- // in one extra buffer being left un-trimmed when it could have
- // been. In general, this is not typical, and we would rather
- // optimized away the TS calculation below for the more common case
- // where PTSes are contiguous.
- bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
- } else {
- // We have no next buffer. Compute the PTS of the frame following
- // the last frame in this buffer by computing the duration of of
- // this frame in media time units and adding it to the PTS of the
- // buffer.
- int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
- / mFrameSize;
-
- if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
- &bufEnd)) {
- ALOGE("Failed to convert frame count of %lld to media time"
- " duration" " (scale factor %d/%u) in %s",
- frameCount,
- mMediaTimeToSampleTransform.a_to_b_numer,
- mMediaTimeToSampleTransform.a_to_b_denom,
- __PRETTY_FUNCTION__);
- break;
- }
- bufEnd += mTimedBufferQueue[trimEnd].pts();
- }
-
- if (bufEnd > mediaTimeNow)
- break;
-
- // Is the buffer we want to use in the middle of a mix operation right
- // now? If so, don't actually trim it. Just wait for the releaseBuffer
- // from the mixer which should be coming back shortly.
- if (!trimEnd && mQueueHeadInFlight) {
- mTrimQueueHeadOnRelease = true;
- }
- }
-
- size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
- if (trimStart < trimEnd) {
- // Update the bookkeeping for framesReady()
- for (size_t i = trimStart; i < trimEnd; ++i) {
- updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
- }
-
- // Now actually remove the buffers from the queue.
- mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
- }
-}
-
-void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
- const char* logTag) {
- ALOG_ASSERT(mTimedBufferQueue.size() > 0,
- "%s called (reason \"%s\"), but timed buffer queue has no"
- " elements to trim.", __FUNCTION__, logTag);
-
- updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
- mTimedBufferQueue.removeAt(0);
-}
-
-void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
- const TimedBuffer& buf,
- const char* logTag __unused) {
- uint32_t bufBytes = buf.buffer()->size();
- uint32_t consumedAlready = buf.position();
-
- ALOG_ASSERT(consumedAlready <= bufBytes,
- "Bad bookkeeping while updating frames pending. Timed buffer is"
- " only %u bytes long, but claims to have consumed %u"
- " bytes. (update reason: \"%s\")",
- bufBytes, consumedAlready, logTag);
-
- uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
- ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
- "Bad bookkeeping while updating frames pending. Should have at"
- " least %u queued frames, but we think we have only %u. (update"
- " reason: \"%s\")",
- bufFrames, mFramesPendingInQueue, logTag);
-
- mFramesPendingInQueue -= bufFrames;
-}
-
-status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
- const sp<IMemory>& buffer, int64_t pts) {
-
- {
- Mutex::Autolock mttLock(mMediaTimeTransformLock);
- if (!mMediaTimeTransformValid)
- return INVALID_OPERATION;
- }
-
- Mutex::Autolock _l(mTimedBufferQueueLock);
-
- uint32_t bufFrames = buffer->size() / mFrameSize;
- mFramesPendingInQueue += bufFrames;
- mTimedBufferQueue.add(TimedBuffer(buffer, pts));
-
- return NO_ERROR;
-}
-
-status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
- const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
-
- ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
- xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
- target);
-
- if (!(target == TimedAudioTrack::LOCAL_TIME ||
- target == TimedAudioTrack::COMMON_TIME)) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock lock(mMediaTimeTransformLock);
- mMediaTimeTransform = xform;
- mMediaTimeTransformTarget = target;
- mMediaTimeTransformValid = true;
-
- return NO_ERROR;
-}
-
-#define min(a, b) ((a) < (b) ? (a) : (b))
-
-// implementation of getNextBuffer for tracks whose buffers have timestamps
-status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts)
-{
- if (pts == AudioBufferProvider::kInvalidPTS) {
- buffer->raw = NULL;
- buffer->frameCount = 0;
- mTimedAudioOutputOnTime = false;
- return INVALID_OPERATION;
- }
-
- Mutex::Autolock _l(mTimedBufferQueueLock);
-
- ALOG_ASSERT(!mQueueHeadInFlight,
- "getNextBuffer called without releaseBuffer!");
-
- while (true) {
-
- // if we have no timed buffers, then fail
- if (mTimedBufferQueue.isEmpty()) {
- buffer->raw = NULL;
- buffer->frameCount = 0;
- return NOT_ENOUGH_DATA;
- }
-
- TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
-
- // calculate the PTS of the head of the timed buffer queue expressed in
- // local time
- int64_t headLocalPTS;
- {
- Mutex::Autolock mttLock(mMediaTimeTransformLock);
-
- ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
-
- if (mMediaTimeTransform.a_to_b_denom == 0) {
- // the transform represents a pause, so yield silence
- timedYieldSilence_l(buffer->frameCount, buffer);
- return NO_ERROR;
- }
-
- int64_t transformedPTS;
- if (!mMediaTimeTransform.doForwardTransform(head.pts(),
- &transformedPTS)) {
- // the transform failed. this shouldn't happen, but if it does
- // then just drop this buffer
- ALOGW("timedGetNextBuffer transform failed");
- buffer->raw = NULL;
- buffer->frameCount = 0;
- trimTimedBufferQueueHead_l("getNextBuffer; no transform");
- return NO_ERROR;
- }
-
- if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
- if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
- &headLocalPTS)) {
- buffer->raw = NULL;
- buffer->frameCount = 0;
- return INVALID_OPERATION;
- }
- } else {
- headLocalPTS = transformedPTS;
- }
- }
-
- uint32_t sr = sampleRate();
-
- // adjust the head buffer's PTS to reflect the portion of the head buffer
- // that has already been consumed
- int64_t effectivePTS = headLocalPTS +
- ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
-
- // Calculate the delta in samples between the head of the input buffer
- // queue and the start of the next output buffer that will be written.
- // If the transformation fails because of over or underflow, it means
- // that the sample's position in the output stream is so far out of
- // whack that it should just be dropped.
- int64_t sampleDelta;
- if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
- ALOGV("*** head buffer is too far from PTS: dropped buffer");
- trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
- " mix");
- continue;
- }
- if (!mLocalTimeToSampleTransform.doForwardTransform(
- (effectivePTS - pts) << 32, &sampleDelta)) {
- ALOGV("*** too late during sample rate transform: dropped buffer");
- trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
- continue;
- }
-
- ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
- " sampleDelta=[%d.%08x]",
- head.pts(), head.position(), pts,
- static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
- + (sampleDelta >> 32)),
- static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
-
- // if the delta between the ideal placement for the next input sample and
- // the current output position is within this threshold, then we will
- // concatenate the next input samples to the previous output
- const int64_t kSampleContinuityThreshold =
- (static_cast<int64_t>(sr) << 32) / 250;
-
- // if this is the first buffer of audio that we're emitting from this track
- // then it should be almost exactly on time.
- const int64_t kSampleStartupThreshold = 1LL << 32;
-
- if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
- (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
- // the next input is close enough to being on time, so concatenate it
- // with the last output
- timedYieldSamples_l(buffer);
-
- ALOGVV("*** on time: head.pos=%d frameCount=%u",
- head.position(), buffer->frameCount);
- return NO_ERROR;
- }
-
- // Looks like our output is not on time. Reset our on timed status.
- // Next time we mix samples from our input queue, then should be within
- // the StartupThreshold.
- mTimedAudioOutputOnTime = false;
- if (sampleDelta > 0) {
- // the gap between the current output position and the proper start of
- // the next input sample is too big, so fill it with silence
- uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
-
- timedYieldSilence_l(framesUntilNextInput, buffer);
- ALOGV("*** silence: frameCount=%u", buffer->frameCount);
- return NO_ERROR;
- } else {
- // the next input sample is late
- uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
- size_t onTimeSamplePosition =
- head.position() + lateFrames * mFrameSize;
-
- if (onTimeSamplePosition > head.buffer()->size()) {
- // all the remaining samples in the head are too late, so
- // drop it and move on
- ALOGV("*** too late: dropped buffer");
- trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
- continue;
- } else {
- // skip over the late samples
- head.setPosition(onTimeSamplePosition);
-
- // yield the available samples
- timedYieldSamples_l(buffer);
-
- ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
- return NO_ERROR;
- }
- }
- }
-}
-
-// Yield samples from the timed buffer queue head up to the given output
-// buffer's capacity.
-//
-// Caller must hold mTimedBufferQueueLock
-void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
- AudioBufferProvider::Buffer* buffer) {
-
- const TimedBuffer& head = mTimedBufferQueue[0];
-
- buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
- head.position());
-
- uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
- mFrameSize);
- size_t framesRequested = buffer->frameCount;
- buffer->frameCount = min(framesLeftInHead, framesRequested);
-
- mQueueHeadInFlight = true;
- mTimedAudioOutputOnTime = true;
-}
-
-// Yield samples of silence up to the given output buffer's capacity
-//
-// Caller must hold mTimedBufferQueueLock
-void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
- uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
-
- // lazily allocate a buffer filled with silence
- if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
- delete [] mTimedSilenceBuffer;
- mTimedSilenceBufferSize = numFrames * mFrameSize;
- mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
- memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
- }
-
- buffer->raw = mTimedSilenceBuffer;
- size_t framesRequested = buffer->frameCount;
- buffer->frameCount = min(numFrames, framesRequested);
-
- mTimedAudioOutputOnTime = false;
-}
-
-// AudioBufferProvider interface
-void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
- AudioBufferProvider::Buffer* buffer) {
-
- Mutex::Autolock _l(mTimedBufferQueueLock);
-
- // If the buffer which was just released is part of the buffer at the head
- // of the queue, be sure to update the amt of the buffer which has been
- // consumed. If the buffer being returned is not part of the head of the
- // queue, its either because the buffer is part of the silence buffer, or
- // because the head of the timed queue was trimmed after the mixer called
- // getNextBuffer but before the mixer called releaseBuffer.
- if (buffer->raw == mTimedSilenceBuffer) {
- ALOG_ASSERT(!mQueueHeadInFlight,
- "Queue head in flight during release of silence buffer!");
- goto done;
- }
-
- ALOG_ASSERT(mQueueHeadInFlight,
- "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
- " head in flight.");
-
- if (mTimedBufferQueue.size()) {
- TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
-
- void* start = head.buffer()->pointer();
- void* end = reinterpret_cast<void*>(
- reinterpret_cast<uint8_t*>(head.buffer()->pointer())
- + head.buffer()->size());
-
- ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
- "released buffer not within the head of the timed buffer"
- " queue; qHead = [%p, %p], released buffer = %p",
- start, end, buffer->raw);
-
- head.setPosition(head.position() +
- (buffer->frameCount * mFrameSize));
- mQueueHeadInFlight = false;
-
- ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
- "Bad bookkeeping during releaseBuffer! Should have at"
- " least %u queued frames, but we think we have only %u",
- buffer->frameCount, mFramesPendingInQueue);
-
- mFramesPendingInQueue -= buffer->frameCount;
-
- if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
- || mTrimQueueHeadOnRelease) {
- trimTimedBufferQueueHead_l("releaseBuffer");
- mTrimQueueHeadOnRelease = false;
- }
- } else {
- LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
- " buffers in the timed buffer queue");
- }
-
-done:
- buffer->raw = 0;
- buffer->frameCount = 0;
-}
-
-size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
- Mutex::Autolock _l(mTimedBufferQueueLock);
- return mFramesPendingInQueue;
-}
-
-AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
- : mPTS(0), mPosition(0) {}
-
-AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
- const sp<IMemory>& buffer, int64_t pts)
- : mBuffer(buffer), mPTS(pts), mPosition(0) {}
-
-
-// ----------------------------------------------------------------------------
-
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
PlaybackThread *playbackThread,
DuplicatingThread *sourceThread,
@@ -1855,7 +1295,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts)
+ AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy");
Proxy::Buffer buf;
@@ -1866,7 +1306,7 @@
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
- status = Track::getNextBuffer(buffer, pts);
+ status = Track::getNextBuffer(buffer);
return status;
}
@@ -2005,8 +1445,7 @@
}
// AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts __unused)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
@@ -2146,7 +1585,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts)
+ AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy");
Proxy::Buffer buf;
@@ -2158,7 +1597,7 @@
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
- status = RecordTrack::getNextBuffer(buffer, pts);
+ status = RecordTrack::getNextBuffer(buffer);
return status;
}
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 7893778..bae3c5b 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -272,9 +272,7 @@
mFrameSize(frameSize),
mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
}
- virtual status_t getNextBuffer(Buffer* buffer,
- int64_t pts = kInvalidPTS) {
- (void)pts; // suppress warning
+ virtual status_t getNextBuffer(Buffer* buffer) {
size_t requestedFrames = buffer->frameCount;
if (requestedFrames > mNumFrames - mNextFrame) {
buffer->frameCount = mNumFrames - mNextFrame;
diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk
index e152468..6182de0 100644
--- a/services/audioflinger/tests/Android.mk
+++ b/services/audioflinger/tests/Android.mk
@@ -47,7 +47,6 @@
LOCAL_SHARED_LIBRARIES := \
libeffects \
libnbaio \
- libcommon_time_client \
libaudioresampler \
libaudioutils \
libdl \
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
index 8da6245..65e22da 100644
--- a/services/audioflinger/tests/test-mixer.cpp
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -307,7 +307,7 @@
(char *) auxAddr + i * auxFrameSize);
}
}
- mixer->process(AudioBufferProvider::kInvalidPTS);
+ mixer->process();
}
outputFrames = i; // reset output frames to the data actually produced.
diff --git a/services/audioflinger/tests/test_utils.h b/services/audioflinger/tests/test_utils.h
index 3d51cdc..283c768 100644
--- a/services/audioflinger/tests/test_utils.h
+++ b/services/audioflinger/tests/test_utils.h
@@ -112,7 +112,7 @@
mNextIdx = 0;
}
- virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS)
+ virtual android::status_t getNextBuffer(Buffer* buffer)
{
size_t requestedFrames = buffer->frameCount;
if (requestedFrames > mNumFrames - mNextFrame) {