Merge "Add multichannel capability to AudioFlinger" into lmp-dev (0f036bec) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioFlinger.cpp

+31 −13

Original line number	Diff line number	Diff line
		@@ -1594,16 +1594,25 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_
		audio_stream_out_t *outStream = NULL;

		// FOR TESTING ONLY:
		// Enable increased sink precision for mixing mode if kEnableExtendedPrecision is true.
		if (kEnableExtendedPrecision && // Check only for Normal Mixing mode
		!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD \| AUDIO_OUTPUT_FLAG_DIRECT))) {
		// Update format
		//config.format = AUDIO_FORMAT_PCM_FLOAT;
		//config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
		//config.format = AUDIO_FORMAT_PCM_32_BIT;
		//config.format = AUDIO_FORMAT_PCM_8_24_BIT;
		// This if statement allows overriding the audio policy settings
		// and forcing a specific format or channel mask to the HAL/Sink device for testing.
		if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD \| AUDIO_OUTPUT_FLAG_DIRECT))) {
		// Check only for Normal Mixing mode
		if (kEnableExtendedPrecision) {
		// Specify format (uncomment one below to choose)
		//config->format = AUDIO_FORMAT_PCM_FLOAT;
		//config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
		//config->format = AUDIO_FORMAT_PCM_32_BIT;
		//config->format = AUDIO_FORMAT_PCM_8_24_BIT;
		// ALOGV("openOutput() upgrading format to %#08x", config.format);
		}
		if (kEnableExtendedChannels) {
		// Specify channel mask (uncomment one below to choose)
		//config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
		//config->channel_mask = audio_channel_mask_from_representation_and_bits(
		// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
		}
		}

		status_t status = hwDevHal->open_output_stream(hwDevHal,
		id,
		@@ -1613,8 +1622,8 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_
		&outStream);

		mHardwareStatus = AUDIO_HW_IDLE;
		ALOGV("openOutput_l() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
		"Channels %x, status %d",
		ALOGV("openOutput() openOutputStream returned output %p, sampleRate %d, Format %#x, "
		"channelMask %#x, status %d",
		outStream,
		config->sample_rate,
		config->format,
		@@ -1630,7 +1639,7 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_
		ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
		} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
		\|\| !isValidPcmSinkFormat(config->format)
		\|\| (config->channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
		\|\| !isValidPcmSinkChannelMask(config->channel_mask)) {
		thread = new DirectOutputThread(this, output, id, device);
		ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
		} else {
		@@ -1792,7 +1801,6 @@ status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
		closeOutputFinish(thread);
		}

		thread.clear();
		return NO_ERROR;
		}

		@@ -2515,6 +2523,16 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId,
		return INVALID_OPERATION;
		}

		// Check whether the destination thread has a channel count of FCC_2, which is
		// currently required for (most) effects. Prevent moving the effect chain here rather
		// than disabling the addEffect_l() call in dstThread below.
		if (dstThread->mChannelCount != FCC_2) {
		ALOGW("moveEffectChain_l() effect chain failed because"
		" destination thread %p channel count(%u) != %u",
		dstThread, dstThread->mChannelCount, FCC_2);
		return INVALID_OPERATION;
		}

		// remove chain first. This is useful only if reconfiguring effect chain on same output thread,
		// so that a new chain is created with correct parameters when first effect is added. This is
		// otherwise unnecessary as removeEffect_l() will remove the chain when last effect is

services/audioflinger/AudioFlinger.h

+26 −1

Original line number	Diff line number	Diff line
		@@ -55,6 +55,7 @@
		#include "FastMixer.h"
		#include <media/nbaio/NBAIO.h>
		#include "AudioWatchdog.h"
		#include "AudioMixer.h"

		#include <powermanager/IPowerManager.h>

		@@ -327,6 +328,30 @@ private:
		audio_devices_t devices);
		void purgeStaleEffects_l();

		// Set kEnableExtendedChannels to true to enable greater than stereo output
		// for the MixerThread and device sink. Number of channels allowed is
		// FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS.
		static const bool kEnableExtendedChannels = false;

		// Returns true if channel mask is permitted for the PCM sink in the MixerThread
		static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) {
		switch (audio_channel_mask_get_representation(channelMask)) {
		case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
		uint32_t channelCount = FCC_2; // stereo is default
		if (kEnableExtendedChannels) {
		channelCount = audio_channel_count_from_out_mask(channelMask);
		if (channelCount > AudioMixer::MAX_NUM_CHANNELS) {
		return false;
		}
		}
		// check that channelMask is the "canonical" one we expect for the channelCount.
		return channelMask == audio_channel_out_mask_from_count(channelCount);
		}
		default:
		return false;
		}
		}

		// Set kEnableExtendedPrecision to true to use extended precision in MixerThread
		static const bool kEnableExtendedPrecision = true;

		@@ -565,7 +590,7 @@ private:
		uint32_t version() const { return mHwDevice->common.version; }

		private:
		audio_module_handle_t mHandle;
		const audio_module_handle_t mHandle;
		const char * const mModuleName;
		audio_hw_device_t * const mHwDevice;
		const Flags mFlags;

services/audioflinger/AudioMixer.cpp

+4 −1

Original line number	Diff line number	Diff line
		@@ -40,6 +40,7 @@
		#include <common_time/cc_helper.h>

		#include <media/EffectsFactoryApi.h>
		#include <audio_effects/effect_downmix.h>

		#include "AudioMixerOps.h"
		#include "AudioMixer.h"
		@@ -941,7 +942,9 @@ bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSam
		// but if none exists, it is the channel count (1 for mono).
		const int resamplerChannelCount = downmixerBufferProvider != NULL
		? mMixerChannelCount : channelCount;
		ALOGVV("Creating resampler with %#x format\n", mMixerInFormat);
		ALOGVV("Creating resampler:"
		" format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
		mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
		resampler = AudioResampler::create(
		mMixerInFormat,
		resamplerChannelCount,

services/audioflinger/AudioMixer.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -26,7 +26,7 @@
		#include <media/AudioBufferProvider.h>
		#include "AudioResampler.h"

		#include <audio_effects/effect_downmix.h>
		#include <hardware/audio_effect.h>
		#include <system/audio.h>
		#include <media/nbaio/NBLog.h>

services/audioflinger/FastMixer.cpp

+24 −5

Original line number	Diff line number	Diff line
		@@ -55,6 +55,7 @@ FastMixer::FastMixer() : FastThread(),
		mixer(NULL),
		mSinkBuffer(NULL),
		mSinkBufferSize(0),
		mSinkChannelCount(FCC_2),
		mMixerBuffer(NULL),
		mMixerBufferSize(0),
		mMixerBufferFormat(AUDIO_FORMAT_PCM_16_BIT),
		@@ -71,6 +72,9 @@ FastMixer::FastMixer() : FastThread(),
		current = &initial;

		mDummyDumpState = &dummyDumpState;
		// TODO: Add channel mask to NBAIO_Format.
		// We assume that the channel mask must be a valid positional channel mask.
		mSinkChannelMask = audio_channel_out_mask_from_count(mSinkChannelCount);

		unsigned i;
		for (i = 0; i < FastMixerState::kMaxFastTracks; ++i) {
		@@ -148,10 +152,17 @@ void FastMixer::onStateChange()
		if (outputSink == NULL) {
		format = Format_Invalid;
		sampleRate = 0;
		mSinkChannelCount = 0;
		mSinkChannelMask = AUDIO_CHANNEL_NONE;
		} else {
		format = outputSink->format();
		sampleRate = Format_sampleRate(format);
		ALOG_ASSERT(Format_channelCount(format) == FCC_2);
		mSinkChannelCount = Format_channelCount(format);
		LOG_ALWAYS_FATAL_IF(mSinkChannelCount > AudioMixer::MAX_NUM_CHANNELS);

		// TODO: Add channel mask to NBAIO_Format
		// We assume that the channel mask must be a valid positional channel mask.
		mSinkChannelMask = audio_channel_out_mask_from_count(mSinkChannelCount);
		}
		dumpState->mSampleRate = sampleRate;
		}
		@@ -169,10 +180,12 @@ void FastMixer::onStateChange()
		// implementation; it would be better to have normal mixer allocate for us
		// to avoid blocking here and to prevent possible priority inversion
		mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks);
		const size_t mixerFrameSize = FCC_2 * audio_bytes_per_sample(mMixerBufferFormat);
		const size_t mixerFrameSize = mSinkChannelCount
		* audio_bytes_per_sample(mMixerBufferFormat);
		mMixerBufferSize = mixerFrameSize * frameCount;
		(void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
		const size_t sinkFrameSize = FCC_2 * audio_bytes_per_sample(format.mFormat);
		const size_t sinkFrameSize = mSinkChannelCount
		* audio_bytes_per_sample(format.mFormat);
		if (sinkFrameSize > mixerFrameSize) { // need a sink buffer
		mSinkBufferSize = sinkFrameSize * frameCount;
		(void)posix_memalign(&mSinkBuffer, 32, mSinkBufferSize);
		@@ -252,6 +265,10 @@ void FastMixer::onStateChange()
		AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
		mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
		(void *)(uintptr_t)fastTrack->mFormat);
		mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
		(void *)(uintptr_t)fastTrack->mChannelMask);
		mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MIXER_CHANNEL_MASK,
		(void *)(uintptr_t)mSinkChannelMask);
		mixer->enable(name);
		}
		generations[i] = fastTrack->mGeneration;
		@@ -287,6 +304,8 @@ void FastMixer::onStateChange()
		(void *)(uintptr_t)fastTrack->mFormat);
		mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
		(void *)(uintptr_t)fastTrack->mChannelMask);
		mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MIXER_CHANNEL_MASK,
		(void *)(uintptr_t)mSinkChannelMask);
		// already enabled
		}
		generations[i] = fastTrack->mGeneration;