Allow AUDIO_FORMAT_PCM_8_BIT AudioTrack buffers
Previously conversion to AUDIO_FORMAT_PCM_16_BIT was required.
Client shared memory for static tracks and AudioFlinger track creation
and buffer delivery now use native 8 bit PCM data.
Change-Id: I485c07a4971fde9a25442bd43fed95019d39abcc
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 99750bd..d4bacc0 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -322,12 +322,6 @@
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
- // AudioFlinger does not currently support 8-bit data in shared memory
- if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
- ALOGE("8-bit data in shared memory is not supported");
- return BAD_VALUE;
- }
-
// force direct flag if format is not linear PCM
// or offload was requested
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
@@ -351,12 +345,9 @@
} else {
mFrameSize = sizeof(uint8_t);
}
- mFrameSizeAF = mFrameSize;
} else {
ALOG_ASSERT(audio_is_linear_pcm(format));
mFrameSize = channelCount * audio_bytes_per_sample(format);
- mFrameSizeAF = channelCount * audio_bytes_per_sample(
- format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format);
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
}
@@ -1045,12 +1036,12 @@
mNotificationFramesAct = frameCount;
}
} else if (mSharedBuffer != 0) {
-
- // Ensure that buffer alignment matches channel count
- // 8-bit data in shared memory is not currently supported by AudioFlinger
- size_t alignment = audio_bytes_per_sample(
- mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat);
+ // FIXME: Ensure client side memory buffers need
+ // not have additional alignment beyond sample
+ // (e.g. 16 bit stereo accessed as 32 bit frame).
+ size_t alignment = audio_bytes_per_sample(mFormat);
if (alignment & 1) {
+ // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
if (mChannelCount > 1) {
@@ -1068,7 +1059,7 @@
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
- frameCount = mSharedBuffer->size() / mFrameSizeAF;
+ frameCount = mSharedBuffer->size() / mFrameSize;
} else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
@@ -1129,10 +1120,7 @@
// but we will still need the original value also
sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
mSampleRate,
- // AudioFlinger only sees 16-bit PCM
- mFormat == AUDIO_FORMAT_PCM_8_BIT &&
- !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ?
- AUDIO_FORMAT_PCM_16_BIT : mFormat,
+ mFormat,
mChannelMask,
&temp,
&trackFlags,
@@ -1256,9 +1244,9 @@
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
- mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
} else {
- mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
mProxy = mStaticProxy;
}
@@ -1378,7 +1366,7 @@
} while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
- audioBuffer->size = buffer.mFrameCount * mFrameSizeAF;
+ audioBuffer->size = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
@@ -1392,7 +1380,7 @@
return;
}
- size_t stepCount = audioBuffer->size / mFrameSizeAF;
+ size_t stepCount = audioBuffer->size / mFrameSize;
if (stepCount == 0) {
return;
}
@@ -1458,14 +1446,8 @@
}
size_t toWrite;
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- // Divide capacity by 2 to take expansion into account
- toWrite = audioBuffer.size >> 1;
- memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite);
- } else {
- toWrite = audioBuffer.size;
- memcpy(audioBuffer.i8, buffer, toWrite);
- }
+ toWrite = audioBuffer.size;
+ memcpy(audioBuffer.i8, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
@@ -1669,7 +1651,7 @@
}
// These fields don't need to be cached, because they are assigned only by set():
- // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags
+ // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
// mFlags is also assigned by createTrack_l(), but not the bit we care about.
mLock.unlock();
@@ -1813,13 +1795,6 @@
}
}
- // Divide buffer size by 2 to take into account the expansion
- // due to 8 to 16 bit conversion: the callback must fill only half
- // of the destination buffer
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- audioBuffer.size >>= 1;
- }
-
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
size_t writtenSize = audioBuffer.size;
@@ -1839,13 +1814,7 @@
return WAIT_PERIOD_MS * 1000000LL;
}
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- // 8 to 16 bit conversion, note that source and destination are the same address
- memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
- audioBuffer.size <<= 1;
- }
-
- size_t releasedFrames = audioBuffer.size / mFrameSizeAF;
+ size_t releasedFrames = audioBuffer.size / mFrameSize;
audioBuffer.frameCount = releasedFrames;
mRemainingFrames -= releasedFrames;
if (misalignment >= releasedFrames) {