Merge "Add client side support for more AUDIO_FORMAT_PCM_*" (96b292fd) · Commits · e / os / android_frameworks_av

include/media/AudioTrack.h

+3 −2

Original line number	Diff line number	Diff line
		@@ -154,8 +154,9 @@ 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 (e.g AUDIO_FORMAT_PCM_16_BIT for signed
		* 16 bits per sample).
		* 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.
		* 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

media/libmedia/AudioTrack.cpp

+19 −7

Original line number	Diff line number	Diff line
		@@ -315,12 +315,20 @@ status_t AudioTrack::set(
		flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
		}

		if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
		if (audio_is_linear_pcm(format)) {
		mFrameSize = channelCount * audio_bytes_per_sample(format);
		mFrameSizeAF = channelCount * sizeof(int16_t);
		} else {
		mFrameSize = sizeof(uint8_t);
		mFrameSizeAF = 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
		}

		// Make copy of input parameter offloadInfo so that in the future:
		@@ -931,7 +939,11 @@ status_t AudioTrack::createTrack_l(size_t epoch)

		// Ensure that buffer alignment matches channel count
		// 8-bit data in shared memory is not currently supported by AudioFlinger
		size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
		size_t alignment = audio_bytes_per_sample(
		mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat);
		if (alignment & 1) {
		alignment = 1;
		}
		if (mChannelCount > 1) {
		// More than 2 channels does not require stronger alignment than stereo
		alignment <<= 1;
		@@ -947,7 +959,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
		// 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()/mChannelCount/sizeof(int16_t);
		frameCount = mSharedBuffer->size() / mFrameSizeAF;

		} else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {