Merge "remove AudioSystem::getOutputSamplingRateForAttr()" into lmp-mr1-dev (990dad99) · Commits · e / os / android_frameworks_av

include/media/AudioSystem.h

+0 −2

Original line number	Diff line number	Diff line
		@@ -99,8 +99,6 @@ public:
		// to be non-zero if status == NO_ERROR
		static status_t getOutputSamplingRate(uint32_t* samplingRate,
		audio_stream_type_t stream);
		static status_t getOutputSamplingRateForAttr(uint32_t* samplingRate,
		const audio_attributes_t *attr);
		static status_t getOutputFrameCount(size_t* frameCount,
		audio_stream_type_t stream);
		static status_t getOutputLatency(uint32_t* latency,

media/libmedia/AudioSystem.cpp

+0 −13

Original line number	Diff line number	Diff line
		@@ -247,19 +247,6 @@ status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream
		return getSamplingRate(output, samplingRate);
		}

		status_t AudioSystem::getOutputSamplingRateForAttr(uint32_t* samplingRate,
		const audio_attributes_t *attr)
		{
		if (attr == NULL) {
		return BAD_VALUE;
		}
		audio_io_handle_t output = getOutputForAttr(attr);
		if (output == 0) {
		return PERMISSION_DENIED;
		}
		return getSamplingRate(output, samplingRate);
		}

		status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
		uint32_t* samplingRate)
		{

media/libmedia/AudioTrack.cpp

+15 −15

Original line number	Diff line number	Diff line
		@@ -304,17 +304,6 @@ status_t AudioTrack::set(
		mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
		}

		status_t status;
		if (sampleRate == 0) {
		status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes);
		if (status != NO_ERROR) {
		ALOGE("Could not get output sample rate for stream type %d; status %d",
		mStreamType, status);
		return status;
		}
		}
		mSampleRate = sampleRate;

		// these below should probably come from the audioFlinger too...
		if (format == AUDIO_FORMAT_DEFAULT) {
		format = AUDIO_FORMAT_PCM_16_BIT;
		@@ -373,6 +362,12 @@ status_t AudioTrack::set(
		// so no need to check for specific PCM formats here
		}

		// sampling rate must be specified for direct outputs
		if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
		return BAD_VALUE;
		}
		mSampleRate = sampleRate;

		// Make copy of input parameter offloadInfo so that in the future:
		// (a) createTrack_l doesn't need it as an input parameter
		// (b) we can support re-creation of offloaded tracks
		@@ -413,7 +408,7 @@ status_t AudioTrack::set(
		}

		// create the IAudioTrack
		status = createTrack_l();
		status_t status = createTrack_l();

		if (status != NO_ERROR) {
		if (mAudioTrackThread != 0) {
		@@ -680,15 +675,18 @@ status_t AudioTrack::setSampleRate(uint32_t rate)
		return INVALID_OPERATION;
		}

		AutoMutex lock(mLock);
		if (mOutput == AUDIO_IO_HANDLE_NONE) {
		return NO_INIT;
		}
		uint32_t afSamplingRate;
		if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {
		if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
		return NO_INIT;
		}
		if (rate == 0 \|\| rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
		return BAD_VALUE;
		}

		AutoMutex lock(mLock);
		mSampleRate = rate;
		mProxy->setSampleRate(rate);

		@@ -963,7 +961,9 @@ status_t AudioTrack::createTrack_l()
		ALOGE("getSamplingRate(output=%d) status %d", output, status);
		goto release;
		}

		if (mSampleRate == 0) {
		mSampleRate = afSampleRate;
		}
		// Client decides whether the track is TIMED (see below), but can only express a preference
		// for FAST. Server will perform additional tests.
		if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !((