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/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 055556f..b51b94f 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -393,8 +393,7 @@
return BAD_VALUE;
}
mSampleRate = sampleRate;
- mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL;
- mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL;
+ mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// Make copy of input parameter offloadInfo so that in the future:
// (a) createTrack_l doesn't need it as an input parameter
@@ -722,7 +721,7 @@
return NO_INIT;
}
// pitch is emulated by adjusting speed and sampleRate
- const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPitch);
+ const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
return BAD_VALUE;
}
@@ -757,10 +756,10 @@
return mSampleRate;
}
-status_t AudioTrack::setPlaybackRate(float speed, float pitch)
+status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
AutoMutex lock(mLock);
- if (speed == mSpeed && pitch == mPitch) {
+ if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
return NO_ERROR;
}
if (mIsTimed || isOffloadedOrDirect_l()) {
@@ -770,32 +769,37 @@
return INVALID_OPERATION;
}
// pitch is emulated by adjusting speed and sampleRate
- const uint32_t effectiveRate = adjustSampleRate(mSampleRate, pitch);
- const float effectiveSpeed = adjustSpeed(speed, pitch);
- const float effectivePitch = adjustPitch(pitch);
+ const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
+ const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
+ const float effectivePitch = adjustPitch(playbackRate.mPitch);
if (effectiveSpeed < AUDIO_TIMESTRETCH_SPEED_MIN
|| effectiveSpeed > AUDIO_TIMESTRETCH_SPEED_MAX
|| effectivePitch < AUDIO_TIMESTRETCH_PITCH_MIN
|| effectivePitch > AUDIO_TIMESTRETCH_PITCH_MAX) {
return BAD_VALUE;
+ //TODO: add function in AudioResamplerPublic.h to check for validity.
}
// Check if the buffer size is compatible.
if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
- ALOGV("setPlaybackRate(%f, %f) failed", speed, pitch);
+ ALOGV("setPlaybackRate(%f, %f) failed", playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
- mSpeed = speed;
- mPitch = pitch;
- mProxy->setPlaybackRate(effectiveSpeed, effectivePitch);
+ mPlaybackRate = playbackRate;
+ mProxy->setPlaybackRate(playbackRate);
+
+ //modify this
+ AudioPlaybackRate playbackRateTemp = playbackRate;
+ playbackRateTemp.mSpeed = effectiveSpeed;
+ playbackRateTemp.mPitch = effectivePitch;
+ mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
return NO_ERROR;
}
-void AudioTrack::getPlaybackRate(float *speed, float *pitch) const
+const AudioPlaybackRate& AudioTrack::getPlaybackRate() const
{
AutoMutex lock(mLock);
- *speed = mSpeed;
- *pitch = mPitch;
+ return mPlaybackRate;
}
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
@@ -1170,7 +1174,8 @@
if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
// for normal tracks precompute the frame count based on speed.
const size_t minFrameCount = calculateMinFrameCount(
- afLatency, afFrameCount, afSampleRate, mSampleRate, mSpeed);
+ afLatency, afFrameCount, afSampleRate, mSampleRate,
+ mPlaybackRate.mSpeed);
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
@@ -1342,11 +1347,15 @@
gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
mProxy->setSendLevel(mSendLevel);
- const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPitch);
- const float effectiveSpeed = adjustSpeed(mSpeed, mPitch);
- const float effectivePitch = adjustPitch(mPitch);
+ const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
+ const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
+ const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
mProxy->setSampleRate(effectiveSampleRate);
- mProxy->setPlaybackRate(effectiveSpeed, effectivePitch);
+
+ AudioPlaybackRate playbackRateTemp = mPlaybackRate;
+ playbackRateTemp.mSpeed = effectiveSpeed;
+ playbackRateTemp.mPitch = effectivePitch;
+ mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setMinimum(mNotificationFramesAct);
mDeathNotifier = new DeathNotifier(this);
@@ -1716,7 +1725,7 @@
// Cache other fields that will be needed soon
uint32_t sampleRate = mSampleRate;
- float speed = mSpeed;
+ float speed = mPlaybackRate.mSpeed;
uint32_t notificationFrames = mNotificationFramesAct;
if (mRefreshRemaining) {
mRefreshRemaining = false;
@@ -2138,7 +2147,7 @@
}
const int64_t deltaTimeUs = timestampTimeUs - mStartUs;
const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
- / ((double)mSampleRate * mSpeed);
+ / ((double)mSampleRate * mPlaybackRate.mSpeed);
if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
// Verify that the counter can't count faster than the sample rate
@@ -2226,7 +2235,7 @@
mChannelCount, mFrameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n",
- mSampleRate, mSpeed, mStatus);
+ mSampleRate, mPlaybackRate.mSpeed, mStatus);
result.append(buffer);
snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency);
result.append(buffer);