Use AudioPlaybackRate to hold TimestretchBufferProvider parameters
Use this struct to handle the parameters for TimestretchBufferProvider all
across the system.
Add stretch mode and fallback mode to TimestretchBuffer Provider.
Change-Id: I19099924a7003c62e48bb6ead56c785cb129fba2
diff --git a/services/audioflinger/BufferProviders.cpp b/services/audioflinger/BufferProviders.cpp
index dcae5e7..77bf4ac 100644
--- a/services/audioflinger/BufferProviders.cpp
+++ b/services/audioflinger/BufferProviders.cpp
@@ -361,25 +361,25 @@
}
TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
- audio_format_t format, uint32_t sampleRate, float speed, float pitch) :
+ audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) :
mChannelCount(channelCount),
mFormat(format),
mSampleRate(sampleRate),
mFrameSize(channelCount * audio_bytes_per_sample(format)),
- mSpeed(speed),
- mPitch(pitch),
mLocalBufferFrameCount(0),
mLocalBufferData(NULL),
mRemaining(0),
- mSonicStream(sonicCreateStream(sampleRate, mChannelCount))
+ mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),
+ mFallbackFailErrorShown(false)
{
- ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f)",
- this, channelCount, format, sampleRate, speed, pitch);
- mBuffer.frameCount = 0;
-
LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,
"TimestretchBufferProvider can't allocate Sonic stream");
- sonicSetSpeed(mSonicStream, speed);
+
+ setPlaybackRate(playbackRate);
+ ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f %d %d)",
+ this, channelCount, format, sampleRate, playbackRate.mSpeed,
+ playbackRate.mPitch, playbackRate.mStretchMode, playbackRate.mFallbackMode);
+ mBuffer.frameCount = 0;
}
TimestretchBufferProvider::~TimestretchBufferProvider()
@@ -423,8 +423,8 @@
// need to fetch more data
const size_t outputDesired = pBuffer->frameCount - mRemaining;
- mBuffer.frameCount = mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
- ? outputDesired : outputDesired * mSpeed + 1;
+ mBuffer.frameCount = mPlaybackRate.mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
+ ? outputDesired : outputDesired * mPlaybackRate.mSpeed + 1;
status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
@@ -491,13 +491,13 @@
mRemaining = 0;
}
-status_t TimestretchBufferProvider::setPlaybackRate(float speed, float pitch)
+status_t TimestretchBufferProvider::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
- mSpeed = speed;
- mPitch = pitch;
-
- sonicSetSpeed(mSonicStream, speed);
+ mPlaybackRate = playbackRate;
+ mFallbackFailErrorShown = false;
+ sonicSetSpeed(mSonicStream, mPlaybackRate.mSpeed);
//TODO: pitch is ignored for now
+ //TODO: optimize: if parameters are the same, don't do any extra computation.
return OK;
}
@@ -508,33 +508,68 @@
// Note dstFrames is the required number of frames.
// Ensure consumption from src is as expected.
- const size_t targetSrc = *dstFrames * mSpeed;
+ //TODO: add logic to track "very accurate" consumption related to speed, original sampling
+ //rate, actual frames processed.
+ const size_t targetSrc = *dstFrames * mPlaybackRate.mSpeed;
if (*srcFrames < targetSrc) { // limit dst frames to that possible
- *dstFrames = *srcFrames / mSpeed;
+ *dstFrames = *srcFrames / mPlaybackRate.mSpeed;
} else if (*srcFrames > targetSrc + 1) {
*srcFrames = targetSrc + 1;
}
- switch (mFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- if (sonicWriteFloatToStream(mSonicStream, (float*)srcBuffer, *srcFrames) != 1) {
- ALOGE("sonicWriteFloatToStream cannot realloc");
- *srcFrames = 0; // cannot consume all of srcBuffer
+ if (mPlaybackRate.mSpeed< TIMESTRETCH_SONIC_SPEED_MIN ||
+ mPlaybackRate.mSpeed > TIMESTRETCH_SONIC_SPEED_MAX ) {
+ //fallback mode
+ if (*dstFrames > 0) {
+ switch(mPlaybackRate.mFallbackMode) {
+ case AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT:
+ if (*dstFrames <= *srcFrames) {
+ size_t copySize = mFrameSize * *dstFrames;
+ memcpy(dstBuffer, srcBuffer, copySize);
+ } else {
+ // cyclically repeat the source.
+ for (size_t count = 0; count < *dstFrames; count += *srcFrames) {
+ size_t remaining = min(*srcFrames, *dstFrames - count);
+ memcpy((uint8_t*)dstBuffer + mFrameSize * count,
+ srcBuffer, mFrameSize * remaining);
+ }
+ }
+ break;
+ case AUDIO_TIMESTRETCH_FALLBACK_DEFAULT:
+ case AUDIO_TIMESTRETCH_FALLBACK_MUTE:
+ memset(dstBuffer,0, mFrameSize * *dstFrames);
+ break;
+ case AUDIO_TIMESTRETCH_FALLBACK_FAIL:
+ default:
+ if(!mFallbackFailErrorShown) {
+ ALOGE("invalid parameters in TimestretchBufferProvider fallbackMode:%d",
+ mPlaybackRate.mFallbackMode);
+ mFallbackFailErrorShown = true;
+ }
+ break;
+ }
}
- *dstFrames = sonicReadFloatFromStream(mSonicStream, (float*)dstBuffer, *dstFrames);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- if (sonicWriteShortToStream(mSonicStream, (short*)srcBuffer, *srcFrames) != 1) {
- ALOGE("sonicWriteShortToStream cannot realloc");
- *srcFrames = 0; // cannot consume all of srcBuffer
+ } else {
+ switch (mFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ if (sonicWriteFloatToStream(mSonicStream, (float*)srcBuffer, *srcFrames) != 1) {
+ ALOGE("sonicWriteFloatToStream cannot realloc");
+ *srcFrames = 0; // cannot consume all of srcBuffer
+ }
+ *dstFrames = sonicReadFloatFromStream(mSonicStream, (float*)dstBuffer, *dstFrames);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ if (sonicWriteShortToStream(mSonicStream, (short*)srcBuffer, *srcFrames) != 1) {
+ ALOGE("sonicWriteShortToStream cannot realloc");
+ *srcFrames = 0; // cannot consume all of srcBuffer
+ }
+ *dstFrames = sonicReadShortFromStream(mSonicStream, (short*)dstBuffer, *dstFrames);
+ break;
+ default:
+ // could also be caught on construction
+ LOG_ALWAYS_FATAL("invalid format %#x for TimestretchBufferProvider", mFormat);
}
- *dstFrames = sonicReadShortFromStream(mSonicStream, (short*)dstBuffer, *dstFrames);
- break;
- default:
- // could also be caught on construction
- LOG_ALWAYS_FATAL("invalid format %#x for TimestretchBufferProvider", mFormat);
}
}
-
// ----------------------------------------------------------------------------
} // namespace android