Merge "Allow AUDIO_FORMAT_PCM_8_BIT AudioTrack buffers" (dda683a2) · Commits · e / os / android_frameworks_av

include/media/AudioTrack.h

+5 −11

Original line number	Diff line number	Diff line
		@@ -79,9 +79,7 @@ public:
		size_t size; // input/output in bytes == frameCount * frameSize
		// on input it is unused
		// on output is the number of bytes actually filled
		// FIXME this is redundant with respect to frameCount,
		// and TRANSFER_OBTAIN mode is broken for 8-bit data
		// since we don't define the frame format
		// FIXME this is redundant with respect to frameCount.

		union {
		void* raw;
		@@ -154,9 +152,9 @@ public:
		* streamType: Select the type of audio stream this track is attached to
		* (e.g. AUDIO_STREAM_MUSIC).
		* sampleRate: Data source sampling rate in Hz.
		* format: Audio format. For mixed tracks, any PCM format supported by server is OK
		* or AUDIO_FORMAT_PCM_8_BIT which is handled on client side. For direct
		* and offloaded tracks, the possible format(s) depends on the output sink.
		* format: Audio format. For mixed tracks, any PCM format supported by server is OK.
		* For direct and offloaded tracks, the possible format(s) depends on the
		* output sink.
		* channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
		* frameCount: Minimum size of track PCM buffer in frames. This defines the
		* application's contribution to the
		@@ -193,7 +191,6 @@ public:

		/* Creates an audio track and registers it with AudioFlinger.
		* With this constructor, the track is configured for static buffer mode.
		* The format must not be 8-bit linear PCM.
		* Data to be rendered is passed in a shared memory buffer
		* identified by the argument sharedBuffer, which must be non-0.
		* The memory should be initialized to the desired data before calling start().
		@@ -701,10 +698,7 @@ protected:
		const audio_offload_info_t* mOffloadInfo;
		audio_attributes_t mAttributes;

		// mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's
		// twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
		size_t mFrameSize; // app-level frame size
		size_t mFrameSizeAF; // AudioFlinger frame size
		size_t mFrameSize; // frame size in bytes

		status_t mStatus;

media/libmedia/AudioTrack.cpp

+15 −46

Original line number	Diff line number	Diff line
		@@ -322,12 +322,6 @@ status_t AudioTrack::set(
		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 @@ status_t AudioTrack::set(
		} 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 @@ status_t AudioTrack::createTrack_l()
		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 @@ status_t AudioTrack::createTrack_l()
		// 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 @@ status_t AudioTrack::createTrack_l()
		// 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 @@ status_t AudioTrack::createTrack_l()
		// 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 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *re
		} 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 @@ void AudioTrack::releaseBuffer(Buffer* audioBuffer)
		return;
		}

		size_t stepCount = audioBuffer->size / mFrameSizeAF;
		size_t stepCount = audioBuffer->size / mFrameSize;
		if (stepCount == 0) {
		return;
		}
		@@ -1458,14 +1446,8 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
		}

		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);
		}
		buffer = ((const char *) buffer) + toWrite;
		userSize -= toWrite;
		written += toWrite;
		@@ -1669,7 +1651,7 @@ nsecs_t AudioTrack::processAudioBuffer()
		}

		// 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 @@ nsecs_t AudioTrack::processAudioBuffer()
		}
		}

		// 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 @@ nsecs_t AudioTrack::processAudioBuffer()
		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) {

services/audioflinger/AudioMixer.h

+10 −4

Original line number	Diff line number	Diff line
		@@ -127,10 +127,16 @@ public:
		size_t getUnreleasedFrames(int name) const;

		static inline bool isValidPcmTrackFormat(audio_format_t format) {
		return format == AUDIO_FORMAT_PCM_16_BIT \|\|
		format == AUDIO_FORMAT_PCM_24_BIT_PACKED \|\|
		format == AUDIO_FORMAT_PCM_32_BIT \|\|
		format == AUDIO_FORMAT_PCM_FLOAT;
		switch (format) {
		case AUDIO_FORMAT_PCM_8_BIT:
		case AUDIO_FORMAT_PCM_16_BIT:
		case AUDIO_FORMAT_PCM_24_BIT_PACKED:
		case AUDIO_FORMAT_PCM_32_BIT:
		case AUDIO_FORMAT_PCM_FLOAT:
		return true;
		default:
		return false;
		}
		}

		private: