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