Add resampler comments and fix a typo

Change-Id: Ie071673875f663de4212eed4a4dff89d51a5a915

4 files changed, 16 insertions, 4 deletions

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 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
         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 @@ public:
     // 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 @@ bool AudioFlinger::RecordThread::threadLoop()
                 } 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 @@ bool AudioFlinger::RecordThread::threadLoop()
                     // 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 @@ bool AudioFlinger::RecordThread::threadLoop()
                     } else {
                         ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
                     }
+                    // now done with mRsmpOutBuffer
 
                 }
                 if (mFramestoDrop == 0) {
@@ -4385,7 +4388,7 @@ void AudioFlinger::RecordThread::readInputParameters()
         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 @@ private:
             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;