Remove TimedAudioTrack and associated code
Bug: 8278435
Change-Id: I095c1a4888e645e14d93b0b15fbef4524a831ca1
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;
}