Use size_t for frame size

except in the control block, where we don't have room.

In AudioFlinger::ThreadBase::TrackBase::getBuffer,
read the frame size from control block only once.

Change-Id: Id6c4bccd4ed3e07d91df6bbea43bae45524f9f4e

7 files changed, 20 insertions, 14 deletions

diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 2fb69b6..84a8f1c 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -206,7 +206,7 @@ public:
             int         channelCount() const;
             int         channels() const;
             uint32_t    frameCount() const;
-            int         frameSize() const;
+            size_t      frameSize() const;
             int         inputSource() const;
 
 
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 60b052b..01a9d05 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -219,7 +219,12 @@ public:
             audio_format_t format() const;
             int         channelCount() const;
             uint32_t    frameCount() const;
-            int         frameSize() const;
+
+    /* Return channelCount * (bit depth per channel / 8).
+     * channelCount is determined from channelMask, and bit depth comes from format.
+     */
+            size_t      frameSize() const;
+
             sp<IMemory>& sharedBuffer();
 
 
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 046d5e9..b661cb9 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -80,7 +80,7 @@ struct audio_track_cblk_t
                 // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
                 // 16 bit because data is converted to 16 bit before being stored in buffer
 
-                uint8_t     frameSize;
+                uint8_t     frameSize;       // would normally be size_t, but 8 bits is plenty
                 uint8_t     pad1;
                 uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
 
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 8e4a9d6..32b5bac 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -263,7 +263,7 @@ uint32_t AudioRecord::frameCount() const
     return mFrameCount;
 }
 
-int AudioRecord::frameSize() const
+size_t AudioRecord::frameSize() const
 {
     if (audio_is_linear_pcm(mFormat)) {
         return channelCount()*audio_bytes_per_sample(mFormat);
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 9c650ad..a66a1dd 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -295,7 +295,7 @@ uint32_t AudioTrack::frameCount() const
     return mCblk->frameCount;
 }
 
-int AudioTrack::frameSize() const
+size_t AudioTrack::frameSize() const
 {
     if (audio_is_linear_pcm(mFormat)) {
         return channelCount()*audio_bytes_per_sample(mFormat);
@@ -979,7 +979,7 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
     ssize_t written = 0;
     const int8_t *src = (const int8_t *)buffer;
     Buffer audioBuffer;
-    size_t frameSz = (size_t)frameSize();
+    size_t frameSz = frameSize();
 
     do {
         audioBuffer.frameCount = userSize/frameSz;
@@ -1137,7 +1137,7 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
 
         audioBuffer.size = writtenSize;
         // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
-        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sampel size of
+        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
         // 16 bit.
         audioBuffer.frameCount = writtenSize/mCblk->frameSize;
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 9fb666e..3f3163e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1118,7 +1118,7 @@ status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args
     result.append(buffer);
     snprintf(buffer, SIZE, "Format: %d\n", mFormat);
     result.append(buffer);
-    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
+    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
     result.append(buffer);
 
     snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
@@ -1727,7 +1727,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters()
     mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
     mChannelCount = (uint16_t)popcount(mChannelMask);
     mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
-    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
+    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
     mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
 
     // FIXME - Current mixer implementation only supports stereo output: Always
@@ -3324,12 +3324,13 @@ uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
 
 void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
     audio_track_cblk_t* cblk = this->cblk();
-    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
-    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
+    size_t frameSize = cblk->frameSize;
+    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
+    int8_t *bufferEnd = bufferStart + frames * frameSize;
 
     // Check validity of returned pointer in case the track control block would have been corrupted.
     if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
-        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
+        ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
         ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
                 server %d, serverBase %d, user %d, userBase %d",
                 bufferStart, bufferEnd, mBuffer, mBufferEnd,
@@ -4800,7 +4801,7 @@ void AudioFlinger::RecordThread::readInputParameters()
     mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
     mChannelCount = (uint16_t)popcount(mChannelMask);
     mFormat = mInput->stream->common.get_format(&mInput->stream->common);
-    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
+    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
     mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
     mFrameCount = mInputBytes / mFrameSize;
     mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index f99e764..cd5f56d 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -536,7 +536,7 @@ private:
                     size_t                  mFrameCount;
                     uint32_t                mChannelMask;
                     uint16_t                mChannelCount;
-                    uint16_t                mFrameSize;
+                    size_t                  mFrameSize;
                     uint32_t                mFormat;
                     Condition               mParamCond;
                     Vector<String8>         mNewParameters;