Clean up frame size in AudioTrack and AudioFlinger
TrackBase::mFrameSize, mChannelMask, and mChannelCount are now const.
Use TrackBase::mFrameSize instead of re-calculating frame size.
AudioFlinger only sees 16-bit PCM format, conversion from 8-bit is
now entirely on the client side. Previously a small part of the
responsibility was on server side also.
size_t is unsigned, so use %u in logs.
Fix theoretical bug where TrackBase constructor was over-allocating space
for non-linear AudioTrack or 8-bit PCM AudioRecord (probably benign).
Change-Id: I7cbbba0bf4dba29ea751d8af341ab8e5cbbdc206
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 99d583d..6fd1b9e 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -520,7 +520,7 @@
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* mBuffers;
- audio_format_t mFormat;
+ audio_format_t mFormat; // as requested by client, not forced to 16-bit
audio_stream_type_t mStreamType;
uint8_t mChannelCount;
uint8_t mMuted;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7480807..5fb36ee 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -858,7 +858,9 @@
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
- format,
+ // AudioFlinger only sees 16-bit PCM
+ format == AUDIO_FORMAT_PCM_8_BIT ?
+ AUDIO_FORMAT_PCM_16_BIT : format,
channelMask,
frameCount,
&trackFlags,
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 6406b6c..10f4410 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -472,6 +472,14 @@
goto Exit;
}
+ // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
+ // and we don't yet support 8.24 or 32-bit PCM
+ if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
+ ALOGE("createTrack() invalid format %d", format);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -2280,7 +2288,7 @@
// mNormalSink below
{
ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
- ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
+ ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
"mFrameCount=%d, mNormalFrameCount=%d",
mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
mNormalFrameCount);
@@ -4190,11 +4198,12 @@
mState(IDLE),
mSampleRate(sampleRate),
mFormat(format),
- mFrameSize(0), // will be set to correct value in constructor
+ mChannelMask(channelMask),
+ mChannelCount(popcount(channelMask)),
+ mFrameSize(audio_is_linear_pcm(format) ?
+ mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
mStepServerFailed(false),
mSessionId(sessionId)
- // mChannelCount
- // mChannelMask
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -4204,8 +4213,7 @@
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
- uint8_t channelCount = popcount(channelMask);
- size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+ size_t bufferSize = frameCount * mFrameSize;
if (sharedBuffer == 0) {
size += bufferSize;
}
@@ -4236,11 +4244,9 @@
// mCblk->server = 0xffff0000;
// mCblk->userBase = 0xffff0000;
// mCblk->serverBase = 0xffff0000;
- mChannelCount = channelCount;
- mChannelMask = channelMask;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
- memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+ memset(mBuffer, 0, bufferSize);
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer (other flags are cleared)
mCblk->flags = CBLK_UNDERRUN;
@@ -4312,17 +4318,16 @@
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
- size_t frameSize = mFrameSize;
- int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
- int8_t *bufferEnd = bufferStart + frames * frameSize;
+ int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
+ int8_t *bufferEnd = bufferStart + frames * mFrameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
"TrackBase::getBuffer buffer out of range:\n"
" start: %p, end %p , mBuffer %p mBufferEnd %p\n"
- " server %u, serverBase %u, user %u, userBase %u, frameSize %d",
+ " server %u, serverBase %u, user %u, userBase %u, frameSize %u",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
- cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize);
+ cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
return bufferStart;
}
@@ -4364,10 +4369,6 @@
mUnderrunCount(0),
mCachedVolume(1.0)
{
- // NOTE: frame size for 8 bit PCM data is based on a sample size of
- // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
- mFrameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) :
- sizeof(uint8_t);
if (mCblk != NULL) {
// to avoid leaking a track name, do not allocate one unless there is an mCblk
mName = thread->getTrackName_l(channelMask, sessionId);
@@ -5411,13 +5412,6 @@
mOverflow(false)
{
ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
- if (format == AUDIO_FORMAT_PCM_16_BIT) {
- mFrameSize = mChannelCount * sizeof(int16_t);
- } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
- mFrameSize = mChannelCount * sizeof(int8_t);
- } else {
- mFrameSize = sizeof(int8_t);
- }
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 8816929..9ddfe28 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -473,14 +473,13 @@
const uint32_t mSampleRate; // initial sample rate only; for tracks which
// support dynamic rates, the current value is in control block
const audio_format_t mFormat;
- size_t mFrameSize; // AudioFlinger's view of frame size in shared memory,
+ const audio_channel_mask_t mChannelMask;
+ const uint8_t mChannelCount;
+ const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory,
// where for AudioTrack (but not AudioRecord),
// 8-bit PCM samples are stored as 16-bit
- // FIXME should be const
bool mStepServerFailed;
const int mSessionId;
- uint8_t mChannelCount;
- audio_channel_mask_t mChannelMask;
Vector < sp<SyncEvent> >mSyncEvents;
};