Add resampler comments and fix a typo
Change-Id: Ie071673875f663de4212eed4a4dff89d51a5a915
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 99e077c..711b62f 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1714,7 +1714,7 @@
AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
// Start record thread
- // RecorThread require both input and output device indication to forward to audio
+ // RecordThread requires both input and output device indication to forward to audio
// pre processing modules
thread = new RecordThread(this,
input,
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 29dc5b6..33e64ce 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -56,6 +56,14 @@
// 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.
+ // A stereo provider delivers a sequence of interleaved pairs of samples.
+ // Multi-channel providers are not supported.
+ // In either case, 'out' holds interleaved pairs of fixed-point signed Q19.12.
+ // That is, for a mono provider, there is an implicit up-channeling.
+ // Since this method accumulates, the caller is responsible for clearing 'out' initially.
+ // FIXME assumes provider is always successful; it should return the actual frame count.
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) = 0;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index f27d908..a9014a2 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -3787,7 +3787,8 @@
} else {
// resampling
- memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+ // resampler accumulates, but we only have one source track
+ memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
// alter output frame count as if we were expecting stereo samples
if (mChannelCount == 1 && mReqChannelCount == 1) {
framesOut >>= 1;
@@ -3797,6 +3798,7 @@
// ditherAndClamp() works as long as all buffers returned by
// mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true.
if (mChannelCount == 2 && mReqChannelCount == 1) {
+ // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
// the resampler always outputs stereo samples:
// do post stereo to mono conversion
@@ -3805,6 +3807,7 @@
} else {
ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
}
+ // now done with mRsmpOutBuffer
}
if (mFramestoDrop == 0) {
@@ -4385,7 +4388,7 @@
mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
mResampler->setSampleRate(mSampleRate);
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
- mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+ mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2];
// optmization: if mono to mono, alter input frame count as if we were inputing
// stereo samples
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 365c790..09cbc5c 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -794,8 +794,9 @@
sp<RecordTrack> mActiveTrack;
Condition mStartStopCond;
AudioResampler *mResampler;
+ // interleaved stereo pairs of fixed-point signed Q19.12
int32_t *mRsmpOutBuffer;
- int16_t *mRsmpInBuffer;
+ int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount]
size_t mRsmpInIndex;
size_t mInputBytes;
const uint32_t mReqChannelCount;