am b9f72ab0: Merge "AudioFlinger: fix repeated underruns for compressed audio" into mnc-dev (703a7fb7) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioStreamOut.cpp

+12 −3

Original line number	Diff line number	Diff line
		@@ -143,9 +143,19 @@ status_t AudioStreamOut::open(
		return status;
		}

		size_t AudioStreamOut::getFrameSize()
		audio_format_t AudioStreamOut::getFormat() const
		{
		return mHalFrameSize;
		return stream->common.get_format(&stream->common);
		}

		uint32_t AudioStreamOut::getSampleRate() const
		{
		return stream->common.get_sample_rate(&stream->common);
		}

		audio_channel_mask_t AudioStreamOut::getChannelMask() const
		{
		return stream->common.get_channels(&stream->common);
		}

		int AudioStreamOut::flush()
		@@ -165,7 +175,6 @@ int AudioStreamOut::standby()
		ALOG_ASSERT(stream != NULL);
		mRenderPosition = 0;
		mFramesWrittenAtStandby = mFramesWritten;
		ALOGI("AudioStreamOut::standby(), mFramesWrittenAtStandby = %llu", mFramesWrittenAtStandby);
		return stream->common.standby(&stream->common);
		}

services/audioflinger/AudioStreamOut.h

+22 −1

Original line number	Diff line number	Diff line
		@@ -75,7 +75,28 @@ public:
		*/
		virtual ssize_t write(const void *buffer, size_t bytes);

		virtual size_t getFrameSize();
		/**
		* @return frame size from the perspective of the application and the AudioFlinger.
		*/
		virtual size_t getFrameSize() const { return mHalFrameSize; }

		/**
		* @return format from the perspective of the application and the AudioFlinger.
		*/
		virtual audio_format_t getFormat() const;

		/**
		* The HAL may be running at a higher sample rate if, for example, playing wrapped EAC3.
		* @return sample rate from the perspective of the application and the AudioFlinger.
		*/
		virtual uint32_t getSampleRate() const;

		/**
		* The HAL is in stereo mode when playing multi-channel compressed audio over HDMI.
		* @return channel mask from the perspective of the application and the AudioFlinger.
		*/
		virtual audio_channel_mask_t getChannelMask() const;


		virtual status_t flush();
		virtual status_t standby();

services/audioflinger/SpdifStreamOut.cpp

+9 −7

Original line number	Diff line number	Diff line
		@@ -37,6 +37,9 @@ SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev,
		audio_format_t format)
		: AudioStreamOut(dev,flags)
		, mSpdifEncoder(this, format)
		, mApplicationFormat(AUDIO_FORMAT_DEFAULT)
		, mApplicationSampleRate(0)
		, mApplicationChannelMask(0)
		{
		}

		@@ -48,6 +51,10 @@ status_t SpdifStreamOut::open(
		{
		struct audio_config customConfig = *config;

		mApplicationFormat = config->format;
		mApplicationSampleRate = config->sample_rate;
		mApplicationChannelMask = config->channel_mask;

		// Some data bursts run at a higher sample rate.
		// TODO Move this into the audio_utils as a static method.
		switch(config->format) {
		@@ -106,20 +113,15 @@ int SpdifStreamOut::standby()
		return AudioStreamOut::standby();
		}

		size_t SpdifStreamOut::getFrameSize()
		{
		return sizeof(int8_t);
		}

		ssize_t SpdifStreamOut::writeDataBurst(const void* buffer, size_t bytes)
		{
		return AudioStreamOut::write(buffer, bytes);
		}

		ssize_t SpdifStreamOut::write(const void* buffer, size_t bytes)
		ssize_t SpdifStreamOut::write(const void* buffer, size_t numBytes)
		{
		// Write to SPDIF wrapper. It will call back to writeDataBurst().
		return mSpdifEncoder.write(buffer, bytes);
		return mSpdifEncoder.write(buffer, numBytes);
		}

		} // namespace android

services/audioflinger/SpdifStreamOut.h

+24 −1

Original line number	Diff line number	Diff line
		@@ -64,7 +64,27 @@ public:
		*/
		virtual ssize_t write(const void* buffer, size_t bytes);

		virtual size_t getFrameSize();
		/**
		* @return frame size from the perspective of the application and the AudioFlinger.
		*/
		virtual size_t getFrameSize() const { return sizeof(int8_t); }

		/**
		* @return format from the perspective of the application and the AudioFlinger.
		*/
		virtual audio_format_t getFormat() const { return mApplicationFormat; }

		/**
		* The HAL may be running at a higher sample rate if, for example, playing wrapped EAC3.
		* @return sample rate from the perspective of the application and the AudioFlinger.
		*/
		virtual uint32_t getSampleRate() const { return mApplicationSampleRate; }

		/**
		* The HAL is in stereo mode when playing multi-channel compressed audio over HDMI.
		* @return channel mask from the perspective of the application and the AudioFlinger.
		*/
		virtual audio_channel_mask_t getChannelMask() const { return mApplicationChannelMask; }

		virtual status_t flush();
		virtual status_t standby();
		@@ -89,6 +109,9 @@ private:
		};

		MySPDIFEncoder mSpdifEncoder;
		audio_format_t mApplicationFormat;
		uint32_t mApplicationSampleRate;
		audio_channel_mask_t mApplicationChannelMask;

		ssize_t writeDataBurst(const void* data, size_t bytes);
		ssize_t writeInternal(const void* buffer, size_t bytes);

services/audioflinger/Threads.cpp

+12 −4

Original line number	Diff line number	Diff line
		@@ -2081,8 +2081,8 @@ int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
		void AudioFlinger::PlaybackThread::readOutputParameters_l()
		{
		// unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
		mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
		mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
		mSampleRate = mOutput->getSampleRate();
		mChannelMask = mOutput->getChannelMask();
		if (!audio_is_output_channel(mChannelMask)) {
		LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
		}
		@@ -2092,8 +2092,12 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
		mChannelMask);
		}
		mChannelCount = audio_channel_count_from_out_mask(mChannelMask);

		// Get actual HAL format.
		mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
		mFormat = mHALFormat;
		// Get format from the shim, which will be different than the HAL format
		// if playing compressed audio over HDMI passthrough.
		mFormat = mOutput->getFormat();
		if (!audio_is_valid_format(mFormat)) {
		LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
		}
		@@ -4559,9 +4563,10 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep
		// app does not call stop() and relies on underrun to stop:
		// hence the test on (track->mRetryCount > 1).
		// If retryCount<=1 then track is about to underrun and be removed.
		// Do not use a high threshold for compressed audio.
		uint32_t minFrames;
		if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()
		&& (track->mRetryCount > 1)) {
		&& (track->mRetryCount > 1) && audio_is_linear_pcm(mFormat)) {
		minFrames = mNormalFrameCount;
		} else {
		minFrames = 1;
		@@ -4650,6 +4655,9 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep
		// it will then automatically call start() when data is available
		android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
		} else if (last) {
		ALOGW("pause because of UNDERRUN, framesReady = %zu,"
		"minFrames = %u, mFormat = %#x",
		track->framesReady(), minFrames, mFormat);
		mixerStatus = MIXER_TRACKS_ENABLED;
		if (mHwSupportsPause && !mHwPaused && !mStandby) {
		doHwPause = true;