DTS Postpro integration (for frameworks/av) (3dc6ae1f) · Commits · e / devices / smdk4412 / android_frameworks_av

include/media/AudioSystem.h

+0 −3

Original line number	Diff line number	Diff line
		@@ -25,9 +25,6 @@
		#include <system/audio_policy.h>
		#include <utils/Errors.h>
		#include <utils/Mutex.h>
		#ifdef QCOM_HARDWARE
		#include <media/AudioParameter.h>
		#endif

		namespace android {

services/audioflinger/Android.mk

+27 −11

Original line number	Diff line number	Diff line
		#
		# This file was modified by DTS, Inc. The portions of the
		# code that are surrounded by "DTS..." are copyrighted and
		# licensed separately, as follows:
		#
		# (C) 2013 DTS, Inc.
		#
		# Licensed under the Apache License, Version 2.0 (the "License");
		# you may not use this file except in compliance with the License.
		# You may obtain a copy of the License at
		#
		# http://www.apache.org/licenses/LICENSE-2.0
		#
		# Unless required by applicable law or agreed to in writing, software
		# distributed under the License is distributed on an "AS IS" BASIS,
		# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		# See the License for the specific language governing permissions and
		# limitations under the License
		#

		LOCAL_PATH:= $(call my-dir)

		include $(CLEAR_VARS)
		@@ -46,14 +66,6 @@ LOCAL_SHARED_LIBRARIES := \
		libdl \
		libpowermanager

		# SRS Processing
		ifeq ($(strip $(BOARD_USES_SRS_TRUEMEDIA)),true)
		LOCAL_SHARED_LIBRARIES += libsrsprocessing
		LOCAL_CFLAGS += -DSRS_PROCESSING
		LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-effects
		endif
		# SRS Processing

		LOCAL_STATIC_LIBRARIES := \
		libscheduling_policy \
		libcpustats \
		@@ -77,7 +89,11 @@ ifeq ($(BOARD_HAVE_PRE_KITKAT_AUDIO_BLOB),true)
		endif

		LOCAL_CFLAGS += -fvisibility=hidden

		ifeq ($(strip $(BOARD_USES_SRS_TRUEMEDIA)),true)
		LOCAL_SHARED_LIBRARIES += libsrsprocessing
		LOCAL_CFLAGS += -DSRS_PROCESSING
		LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-effects
		endif
		include $(BUILD_SHARED_LIBRARY)

		#

services/audioflinger/AudioFlinger.cpp

+28 −18

Original line number	Diff line number	Diff line
		@@ -14,6 +14,24 @@
		** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		** See the License for the specific language governing permissions and
		** limitations under the License.
		**
		** This file was modified by DTS, Inc. The portions of the
		** code that are surrounded by "DTS..." are copyrighted and
		** licensed separately, as follows:
		**
		** (C) 2013 DTS, Inc.
		**
		** Licensed under the Apache License, Version 2.0 (the "License");
		** you may not use this file except in compliance with the License.
		** You may obtain a copy of the License at
		**
		** http://www.apache.org/licenses/LICENSE-2.0
		**
		** Unless required by applicable law or agreed to in writing, software
		** distributed under the License is distributed on an "AS IS" BASIS,
		** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		** See the License for the specific language governing permissions and
		** limitations under the License
		*/


		@@ -63,10 +81,8 @@
		#include <media/nbaio/PipeReader.h>
		#include <media/AudioParameter.h>
		#include <private/android_filesystem_config.h>

		#ifdef SRS_PROCESSING
		#include "srs_processing.h"
		#include "postpro_patch_ics.h"
		#include "postpro_patch.h"
		#endif

		// ----------------------------------------------------------------------------
		@@ -1067,12 +1083,15 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
		// ioHandle == 0 means the parameters are global to the audio hardware interface
		if (ioHandle == 0) {
		Mutex::Autolock _l(mLock);
		status_t final_result = NO_ERROR;

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_ICS_PARAMS_SET(keyValuePairs);
		if (!mDirectAudioTracks.isEmpty())
		audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
		POSTPRO_PATCH_PARAMS_SET(keyValuePairs);
		for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
		PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
		thread->setPostPro();
		}
		#endif
		status_t final_result = NO_ERROR;
		{
		AutoMutex lock(mHardwareLock);
		mHardwareStatus = AUDIO_HW_SET_PARAMETER;
		@@ -1150,12 +1169,6 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
		int device;
		if (param.getInt(key, device) == NO_ERROR) {
		mDirectDevice = device;
		#ifdef SRS_PROCESSING
		ALOGV("setParameters:: routing change to device %d", device);
		POSTPRO_PATCH_ICS_OUTPROC_MIX_ROUTE(desc->trackRefPtr, param, device);
		if(desc->flag & AUDIO_OUTPUT_FLAG_TUNNEL)
		audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
		#endif
		if(mLPAEffectChain != NULL){
		mLPAEffectChain->setDevice_l(device);
		audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
		@@ -1202,8 +1215,9 @@ String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& k
		String8 out_s8;

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_ICS_PARAMS_GET(keys, out_s8);
		POSTPRO_PATCH_PARAMS_GET(keys, out_s8);
		#endif

		for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
		char *s;
		{
		@@ -1923,10 +1937,6 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
		if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
		ALOGI("Using module %d has the primary audio interface", module);
		mPrimaryHardwareDev = outHwDev;
		#ifdef SRS_PROCESSING
		SRS_Processing::RawDataSet(NULL, "qdsp hook", &mPrimaryHardwareDev,
		sizeof(&mPrimaryHardwareDev));
		#endif
		AutoMutex lock(mHardwareLock);
		mHardwareStatus = AUDIO_HW_SET_MODE;
		hwDevHal->set_mode(hwDevHal, mMode);

services/audioflinger/Effects.cpp

+0 −19

Original line number	Diff line number	Diff line
		@@ -30,11 +30,6 @@
		#include "AudioFlinger.h"
		#include "ServiceUtilities.h"

		#ifdef SRS_PROCESSING
		#include "srs_processing.h"
		#include "postpro_patch_ics.h"
		#endif

		// ----------------------------------------------------------------------------

		// Note: the following macro is used for extremely verbose logging message. In
		@@ -1908,9 +1903,6 @@ bool AudioFlinger::applyEffectsOn(void token, int16_t inBuffer,
		mIsEffectConfigChanged = false;

		volatile size_t numEffects = 0;
		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_ICS_OUTPROC_DIRECT_SAMPLES(token, AUDIO_FORMAT_PCM_16_BIT, outBuffer, size, mLPASampleRate, mLPANumChannels);
		#endif

		if(mLPAEffectChain != NULL) {
		numEffects = mLPAEffectChain->getNumEffects();
		@@ -2035,17 +2027,6 @@ void AudioFlinger::DirectAudioTrack::EffectsThreadEntry() {
		break;
		}
		}
		#ifdef SRS_PROCESSING
		} else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) {
		ALOGV("applying effects for TUNNEL");
		char buffer[2];
		//dummy buffer to ensure the SRS processing takes place
		// The API mandates Sample rate and channel mode. Hence
		// defaulted the sample rate channel mode to 48000 and 2 respectively
		POSTPRO_PATCH_ICS_OUTPROC_DIRECT_SAMPLES(static_cast<void *>(this),
		AUDIO_FORMAT_PCM_16_BIT,
		(int16_t*)buffer, 2, 48000, 2);
		#endif
		}
		}
		mEffectLock.unlock();

services/audioflinger/Threads.cpp

+84 −41

Original line number	Diff line number	Diff line
		@@ -14,6 +14,24 @@
		** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		** See the License for the specific language governing permissions and
		** limitations under the License.
		**
		** This file was modified by DTS, Inc. The portions of the
		** code that are surrounded by "DTS..." are copyrighted and
		** licensed separately, as follows:
		**
		** (C) 2013 DTS, Inc.
		**
		** Licensed under the Apache License, Version 2.0 (the "License");
		** you may not use this file except in compliance with the License.
		** You may obtain a copy of the License at
		**
		** http://www.apache.org/licenses/LICENSE-2.0
		**
		** Unless required by applicable law or agreed to in writing, software
		** distributed under the License is distributed on an "AS IS" BASIS,
		** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		** See the License for the specific language governing permissions and
		** limitations under the License
		*/


		@@ -66,8 +84,7 @@
		#endif

		#ifdef SRS_PROCESSING
		#include "srs_processing.h"
		#include "postpro_patch_ics.h"
		#include "postpro_patch.h"
		#endif

		// ----------------------------------------------------------------------------
		@@ -1487,6 +1504,13 @@ float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) con
		return mStreamTypes[stream].volume;
		}

		void AudioFlinger::PlaybackThread::setPostPro()
		{
		Mutex::Autolock _l(mLock);
		if (mType == OFFLOAD)
		broadcast_l();
		}

		// addTrack_l() must be called with ThreadBase::mLock held
		status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
		{
		@@ -2217,13 +2241,6 @@ bool AudioFlinger::PlaybackThread::threadLoop()

		standbyTime = systemTime();

		#ifdef SRS_PROCESSING
		if (mType == MIXER) {
		POSTPRO_PATCH_ICS_OUTPROC_MIX_INIT(this, gettid());
		} else if (mType == DUPLICATING) {
		POSTPRO_PATCH_ICS_OUTPROC_DUPE_INIT(this, gettid());
		}
		#endif
		// MIXER
		nsecs_t lastWarning = 0;

		@@ -2245,6 +2262,18 @@ if (mType == MIXER) {

		acquireWakeLock();

		#ifdef SRS_PROCESSING
		if (mType == MIXER) {
		POSTPRO_PATCH_OUTPROC_PLAY_INIT(this, myName);
		} else if (mType == DUPLICATING) {
		POSTPRO_PATCH_OUTPROC_DUPE_INIT(this, myName);
		} else if (mType == OFFLOAD) {
		POSTPRO_PATCH_OUTPROC_DIRECT_INIT(this, myName);
		} else if (mType == DIRECT) {
		POSTPRO_PATCH_OUTPROC_DIRECT_INIT(this, myName);
		}
		#endif

		// mNBLogWriter->log can only be called while thread mutex mLock is held.
		// So if you need to log when mutex is unlocked, set logString to a non-NULL string,
		// and then that string will be logged at the next convenient opportunity.
		@@ -2403,6 +2432,13 @@ if (mType == MIXER) {
		for (size_t i = 0; i < effectChains.size(); i ++) {
		effectChains[i]->process_l();
		}
		#ifdef SRS_PROCESSING
		// Offload thread
		if (mType == OFFLOAD) {
		char buffer[2];
		POSTPRO_PATCH_OUTPROC_DIRECT_SAMPLES(this, AUDIO_FORMAT_PCM_16_BIT, (int16_t *) buffer, 2, 48000, 2);
		}
		#endif
		}

		// enable changes in effect chain
		@@ -2413,12 +2449,14 @@ if (mType == MIXER) {
		if (sleepTime == 0) {
		#ifdef SRS_PROCESSING
		if (mType == MIXER) {
		POSTPRO_PATCH_ICS_OUTPROC_MIX_SAMPLES(this, mFormat, mMixBuffer,
		mixBufferSize, mSampleRate, mChannelCount);
		POSTPRO_PATCH_OUTPROC_PLAY_SAMPLES(this, mFormat, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount);
		} else if (mType == DUPLICATING) {
		POSTPRO_PATCH_ICS_OUTPROC_DUPE_SAMPLES(this, mFormat, mMixBuffer,
		mixBufferSize, mSampleRate, mChannelCount);
		}
		POSTPRO_PATCH_OUTPROC_DUPE_SAMPLES(this, mFormat, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount);
		} /* else if (mType == OFFLOAD) {
		POSTPRO_PATCH_OUTPROC_DIRECT_SAMPLES(this, mFormat, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount);
		} else if (mType == DIRECT) {
		POSTPRO_PATCH_OUTPROC_DIRECT_SAMPLES(this, mFormat, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount);
		} */
		#endif
		if (mBytesRemaining) {
		ssize_t ret = threadLoop_write();
		@@ -2480,12 +2518,15 @@ if (mType == MIXER) {
		mOutput->stream->common.standby(&mOutput->stream->common);
		}
		}

		#ifdef SRS_PROCESSING
		if (mType == MIXER) {
		POSTPRO_PATCH_ICS_OUTPROC_MIX_EXIT(this, gettid());
		POSTPRO_PATCH_OUTPROC_PLAY_EXIT(this, myName);
		} else if (mType == DUPLICATING) {
		POSTPRO_PATCH_ICS_OUTPROC_DUPE_EXIT(this, gettid());
		POSTPRO_PATCH_OUTPROC_DUPE_EXIT(this, myName);
		} else if (mType == OFFLOAD) {
		POSTPRO_PATCH_OUTPROC_DIRECT_EXIT(this, myName);
		} else if (mType == DIRECT) {
		POSTPRO_PATCH_OUTPROC_DIRECT_EXIT(this, myName);
		}
		#endif
		releaseWakeLock();
		@@ -3459,9 +3500,8 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l()
		String8 keyValuePair = mNewParameters[0];
		AudioParameter param = AudioParameter(keyValuePair);
		int value;

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_ICS_OUTPROC_MIX_ROUTE(this, param, value);
		POSTPRO_PATCH_OUTPROC_PLAY_ROUTE(this, param, value);
		#endif
		if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
		reconfig = true;
		@@ -3859,6 +3899,10 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
		AudioParameter param = AudioParameter(keyValuePair);
		int value;

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_OUTPROC_PLAY_ROUTE(this, param, value);
		#endif

		if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
		// do not accept frame count changes if tracks are open as the track buffer
		// size depends on frame count and correct behavior would not be garantied
		@@ -4546,6 +4590,10 @@ bool AudioFlinger::RecordThread::threadLoop()
		acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1);
		}

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_INPROC_INIT(this, gettid(), mFormat);
		#endif

		// used to verify we've read at least once before evaluating how many bytes were read
		bool readOnce = false;

		@@ -4683,6 +4731,9 @@ bool AudioFlinger::RecordThread::threadLoop()
		InputBytes);
		#else
		mBufferSize);
		#endif
		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_INPROC_SAMPLES(this, mFormat, readInto, mBytesRead, mSampleRate, mChannelCount);
		#endif
		if (mBytesRead <= 0) {
		if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
		@@ -4785,6 +4836,10 @@ bool AudioFlinger::RecordThread::threadLoop()
		mStartStopCond.broadcast();
		}

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_INPROC_EXIT(this, gettid(), mFormat);
		#endif

		releaseWakeLock();

		ALOGV("RecordThread %p exiting", this);
		@@ -5166,6 +5221,10 @@ status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer*

		if (framesReady == 0) {
		mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize);

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_INPROC_SAMPLES(this, mFormat, mRsmpInBuffer, mBytesRead, mSampleRate, mChannelCount);
		#endif
		if (mBytesRead <= 0) {
		if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
		ALOGE("RecordThread::getNextBuffer() Error reading audio input");
		@@ -5216,6 +5275,9 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
		uint32_t reqSamplingRate = mReqSampleRate;
		uint32_t reqChannelCount = mReqChannelCount;

		#ifdef SRS_PROCESSING
		POSTPRO_PATCH_INPROC_ROUTE(this, param, value);
		#endif
		if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
		reqSamplingRate = value;
		reconfig = true;
		@@ -5498,20 +5560,13 @@ AudioFlinger::DirectAudioTrack::DirectAudioTrack(const sp<AudioFlinger>& audioFl
		mClient(client), mEffectConfigChanged(false), mKillEffectsThread(false), mFlag(outflag),
		mEffectsThreadScratchBuffer(NULL)
		{
		#ifdef SRS_PROCESSING
		ALOGD("SRS_Processing - DirectAudioTrack - OutNotify_Init: %p TID %d\n", this, gettid());
		POSTPRO_PATCH_ICS_OUTPROC_DIRECT_INIT(this, gettid());
		SRS_Processing::ProcessOutRoute(SRS_Processing::AUTO, this, outputDesc->device);
		#endif
		if (mFlag & AUDIO_OUTPUT_FLAG_LPA) {
		ALOGV("create effects thread for LPA");
		createEffectThread();
		allocateBufPool();
		#ifdef SRS_PROCESSING
		} else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) {
		ALOGV("create effects thread for TUNNEL");
		createEffectThread();
		#endif
		}
		outputDesc->mVolumeScale = 1.0;
		mDeathRecipient = new PMDeathRecipient(this);
		@@ -5523,28 +5578,16 @@ void AudioFlinger::DirectAudioTrack::signalEffect() {
		mEffectConfigChanged = true;
		mEffectCv.signal();
		}
		#ifdef SRS_PROCESSING
		if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL){
		mEffectConfigChanged = true;
		mEffectCv.signal();
		}
		#endif
		}

		AudioFlinger::DirectAudioTrack::~DirectAudioTrack() {
		#ifdef SRS_PROCESSING
		ALOGD("SRS_Processing - DirectAudioTrack - OutNotify_Init: %p TID %d\n", this, gettid());
		POSTPRO_PATCH_ICS_OUTPROC_DIRECT_EXIT(this, gettid());
		#endif
		if (mFlag & AUDIO_OUTPUT_FLAG_LPA) {
		requestAndWaitForEffectsThreadExit();
		mAudioFlinger->deleteEffectSession();
		deallocateBufPool();
		#ifdef SRS_PROCESSING
		} else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) {
		requestAndWaitForEffectsThreadExit();
		mAudioFlinger->deleteEffectSession();
		#endif
		}
		AudioSystem::releaseOutput(mOutput);
		releaseWakeLock();