Merge "u8500: add ST Ericsson audio configs" into ics (66155a8b) · Commits · e / os / android_frameworks_base

include/media/IAudioFlinger.h

+13 −1

Original line number	Diff line number	Diff line
		@@ -151,13 +151,22 @@ public:
		virtual status_t suspendOutput(int output) = 0;
		virtual status_t restoreOutput(int output) = 0;

		#ifdef STE_AUDIO
		virtual uint32_t *addInputClient(uint32_t clientId) = 0;
		virtual status_t removeInputClient(uint32_t *pClientId) = 0;
		#endif
		virtual int openInput(uint32_t *pDevices,
		uint32_t *pSamplingRate,
		uint32_t *pFormat,
		uint32_t *pChannels,
		#ifdef STE_AUDIO
		uint32_t acoustics,
		uint32_t *pInputClientId = NULL) = 0;
		virtual status_t closeInput(int input, uint32_t* inputClientId = NULL) = 0;
		#else
		uint32_t acoustics) = 0;
		virtual status_t closeInput(int input) = 0;

		#endif
		virtual status_t setStreamOutput(uint32_t stream, int output) = 0;

		virtual status_t setVoiceVolume(float volume) = 0;
		@@ -191,6 +200,9 @@ public:
		#ifdef WITH_QCOM_LPA
		virtual status_t deregisterClient(const sp<IAudioFlingerClient>& client) { return false; };
		#endif
		#ifdef STE_AUDIO
		virtual size_t readInput(uint32_t input, uint32_t inputClientId, void buffer, uint32_t bytes, uint32_t *pOverwrittenBytes) = 0;
		#endif
		};

include/media/IAudioPolicyService.h

+5 −0

Original line number	Diff line number	Diff line
		@@ -75,7 +75,12 @@ public:
		uint32_t format = AUDIO_FORMAT_DEFAULT,
		uint32_t channels = 0,
		audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0,
		#ifdef STE_AUDIO
		int audioSession = 0,
		audio_input_clients *inputClientId = NULL) = 0;
		#else
		int audioSession = 0) = 0;
		#endif
		virtual status_t startInput(audio_io_handle_t input) = 0;
		virtual status_t stopInput(audio_io_handle_t input) = 0;
		virtual void releaseInput(audio_io_handle_t input) = 0;

media/libmedia/IAudioFlinger.cpp

+100 −2

Original line number	Diff line number	Diff line
		@@ -67,6 +67,10 @@ enum {
		#endif
		SUSPEND_OUTPUT,
		RESTORE_OUTPUT,
		#ifdef STE_AUDIO
		ADD_INPUT_CLIENT,
		REMOVE_INPUT_CLIENT,
		#endif
		OPEN_INPUT,
		CLOSE_INPUT,
		SET_STREAM_OUTPUT,
		@@ -81,6 +85,9 @@ enum {
		GET_EFFECT_DESCRIPTOR,
		CREATE_EFFECT,
		MOVE_EFFECTS,
		#ifdef STE_AUDIO
		READ_INPUT,
		#endif
		#ifdef WITH_QCOM_LPA
		SET_FM_VOLUME,
		CREATE_SESSION,
		@@ -567,11 +574,35 @@ public:
		return reply.readInt32();
		}

		#ifdef STE_AUDIO
		virtual uint32_t *addInputClient(uint32_t clientId)
		{
		Parcel data, reply;
		data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
		data.writeInt32(clientId);
		remote()->transact(ADD_INPUT_CLIENT, data, &reply);
		return (uint32_t*) reply.readIntPtr();
		}

		virtual status_t removeInputClient(uint32_t *pClientId)
		{
		Parcel data, reply;
		data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
		data.writeIntPtr((intptr_t)pClientId);
		remote()->transact(REMOVE_INPUT_CLIENT, data, &reply);
		return reply.readInt32();
		}
		#endif
		virtual int openInput(uint32_t *pDevices,
		uint32_t *pSamplingRate,
		uint32_t *pFormat,
		uint32_t *pChannels,
		#ifdef STE_AUDIO
		uint32_t acoustics,
		uint32_t *pInputClientId)
		#else
		uint32_t acoustics)
		#endif
		{
		Parcel data, reply;
		uint32_t devices = pDevices ? *pDevices : 0;
		@@ -585,6 +616,9 @@ public:
		data.writeInt32(format);
		data.writeInt32(channels);
		data.writeInt32(acoustics);
		#ifdef STE_AUDIO
		data.writeIntPtr((intptr_t)pInputClientId);
		#endif
		remote()->transact(OPEN_INPUT, data, &reply);
		int input = reply.readInt32();
		devices = reply.readInt32();
		@@ -598,6 +632,17 @@ public:
		return input;
		}

		#ifdef STE_AUDIO
		virtual status_t closeInput(int input, uint32_t *inputClientId)
		{
		Parcel data, reply;
		data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
		data.writeInt32(input);
		data.writeIntPtr((intptr_t) inputClientId);
		remote()->transact(CLOSE_INPUT, data, &reply);
		return reply.readInt32();
		}
		#else
		virtual status_t closeInput(int input)
		{
		Parcel data, reply;
		@@ -606,7 +651,7 @@ public:
		remote()->transact(CLOSE_INPUT, data, &reply);
		return reply.readInt32();
		}

		#endif
		virtual status_t setStreamOutput(uint32_t stream, int output)
		{
		Parcel data, reply;
		@@ -655,6 +700,20 @@ public:
		return reply.readInt32();
		}

		#ifdef STE_AUDIO
		virtual size_t readInput(uint32_t input, uint32_t inputClientId, void buffer, uint32_t bytes, uint32_t *pOverwrittenBytes)
		{
		Parcel data, reply;
		data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
		data.writeIntPtr((intptr_t) input);
		data.writeInt32(inputClientId);
		data.writeIntPtr((intptr_t) buffer);
		data.writeInt32(bytes);
		data.writeIntPtr((intptr_t) pOverwrittenBytes);
		remote()->transact(READ_INPUT, data, &reply);
		return reply.readInt32();
		}
		#endif
		virtual int newAudioSessionId()
		{
		Parcel data, reply;
		@@ -1092,6 +1151,20 @@ status_t BnAudioFlinger::onTransact(
		reply->writeInt32(restoreOutput(data.readInt32()));
		return NO_ERROR;
		} break;
		#ifdef STE_AUDIO
		case ADD_INPUT_CLIENT: {
		CHECK_INTERFACE(IAudioFlinger, data, reply);
		uint32_t clientId = data.readInt32();
		reply->writeIntPtr((intptr_t)addInputClient(clientId));
		return NO_ERROR;
		} break;
		case REMOVE_INPUT_CLIENT: {
		CHECK_INTERFACE(IAudioFlinger, data, reply);
		uint32_t pClientId = (uint32_t) data.readIntPtr();
		reply->writeInt32(removeInputClient(pClientId));
		return NO_ERROR;
		} break;
		#endif
		case OPEN_INPUT: {
		CHECK_INTERFACE(IAudioFlinger, data, reply);
		uint32_t devices = data.readInt32();
		@@ -1099,12 +1172,19 @@ status_t BnAudioFlinger::onTransact(
		uint32_t format = data.readInt32();
		uint32_t channels = data.readInt32();
		uint32_t acoutics = data.readInt32();

		#ifdef STE_AUDIO
		uint32_t inputClientId = (uint32_t) data.readIntPtr();
		#endif
		int input = openInput(&devices,
		&samplingRate,
		&format,
		&channels,
		#ifdef STE_AUDIO
		acoutics,
		inputClientId);
		#else
		acoutics);
		#endif
		reply->writeInt32(input);
		reply->writeInt32(devices);
		reply->writeInt32(samplingRate);
		@@ -1114,7 +1194,13 @@ status_t BnAudioFlinger::onTransact(
		} break;
		case CLOSE_INPUT: {
		CHECK_INTERFACE(IAudioFlinger, data, reply);
		#ifdef STE_AUDIO
		uint32_t input = data.readInt32();
		uint32_t inputClientId = (uint32_t) data.readIntPtr();
		reply->writeInt32(closeInput(input, inputClientId));
		#else
		reply->writeInt32(closeInput(data.readInt32()));
		#endif
		return NO_ERROR;
		} break;
		case SET_STREAM_OUTPUT: {
		@@ -1227,6 +1313,18 @@ status_t BnAudioFlinger::onTransact(
		reply->writeInt32(moveEffects(session, srcOutput, dstOutput));
		return NO_ERROR;
		} break;
		#ifdef STE_AUDIO
		case READ_INPUT: {
		CHECK_INTERFACE(IAudioFlinger, data, reply);
		uint32_t* input = (uint32_t*) data.readIntPtr();
		uint32_t inputClientId = data.readInt32();
		void* buffer = (void*) data.readIntPtr();
		uint32_t bytes = data.readInt32();
		uint32_t pOverwrittenBytes = (uint32_t) data.readIntPtr();
		reply->writeInt32(readInput(input, inputClientId, buffer, bytes, pOverwrittenBytes));
		return NO_ERROR;
		} break;
		#endif
		default:
		return BBinder::onTransact(code, data, reply, flags);
		}

media/libmedia/IAudioPolicyService.cpp

+17 −0

Original line number	Diff line number	Diff line
		@@ -238,7 +238,12 @@ public:
		uint32_t format,
		uint32_t channels,
		audio_in_acoustics_t acoustics,
		#ifdef STE_AUDIO
		int audioSession,
		audio_input_clients *inputClientId)
		#else
		int audioSession)
		#endif
		{
		Parcel data, reply;
		data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
		@@ -247,6 +252,9 @@ public:
		data.writeInt32(static_cast <uint32_t>(format));
		data.writeInt32(channels);
		data.writeInt32(static_cast <uint32_t>(acoustics));
		#ifdef STE_AUDIO
		data.writeIntPtr((intptr_t)inputClientId);
		#endif
		data.writeInt32(audioSession);
		remote()->transact(GET_INPUT, data, &reply);
		return static_cast <audio_io_handle_t> (reply.readInt32());
		@@ -569,13 +577,22 @@ status_t BnAudioPolicyService::onTransact(
		uint32_t channels = data.readInt32();
		audio_in_acoustics_t acoustics =
		static_cast <audio_in_acoustics_t>(data.readInt32());
		#ifdef STE_AUDIO
		audio_input_clients *inputClientId =
		(audio_input_clients*) data.readIntPtr();
		#endif
		int audioSession = data.readInt32();
		audio_io_handle_t input = getInput(inputSource,
		samplingRate,
		format,
		channels,
		acoustics,
		#ifdef STE_AUDIO
		audioSession,
		inputClientId);
		#else
		audioSession);
		#endif
		reply->writeInt32(static_cast <int>(input));
		return NO_ERROR;
		} break;

services/audioflinger/AudioFlinger.cpp

+116 −4

Original line number	Diff line number	Diff line
		@@ -233,6 +233,7 @@ void AudioFlinger::onFirstRef()
		mHardwareStatus = AUDIO_HW_IDLE;
		}
		}

		}

		status_t AudioFlinger::initCheck() const
		@@ -266,6 +267,9 @@ AudioFlinger::~AudioFlinger()
		audio_hw_device_close(dev);
		}
		mAudioHwDevs.clear();
		#ifdef STE_AUDIO
		delete mInputFMStream;
		#endif
		}

		audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
		@@ -1144,6 +1148,17 @@ unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
		return 0;
		}

		#ifdef STE_AUDIO
		size_t AudioFlinger::readInput(uint32_t input, uint32_t inputClientId, void buffer, uint32_t bytes, uint32_t *pOverwrittenBytes)
		{
		if (input == NULL \|\| buffer == NULL) {
		return 0;
		}

		AudioStreamIn* stream = (AudioStreamIn*)input;
		return 0; //return stream->read(buffer, bytes);
		}
		#endif
		status_t AudioFlinger::setVoiceVolume(float value)
		{
		status_t ret = initCheck();
		@@ -4643,7 +4658,12 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
		uint32_t sampleRate,
		uint32_t channels,
		int id,
		#ifdef STE_AUDIO
		uint32_t device,
		audio_input_clients pInputClientId) :
		#else
		uint32_t device) :
		#endif
		ThreadBase(audioFlinger, id, device),
		mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
		{
		@@ -4653,6 +4673,9 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,

		mReqChannelCount = popcount(channels);
		mReqSampleRate = sampleRate;
		#ifdef STE_AUDIO
		mInputClientId = pInputClientId;
		#endif
		readInputParameters();
		}

		@@ -4787,10 +4810,27 @@ bool AudioFlinger::RecordThread::threadLoop()
		if (framesOut && mFrameCount == mRsmpInIndex) {
		if (framesOut == mFrameCount &&
		((int)mChannelCount == mReqChannelCount \|\| mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
		#ifdef STE_AUDIO
		mBytesRead = mAudioFlinger->readInput((uint32_t*) mInput,
		(uint32_t)mInputClientId,
		buffer.raw,
		mInputBytes,
		NULL);
		#else
		mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
		#endif
		framesOut = 0;
		} else {
		#ifdef STE_AUDIO
		mBytesRead = mAudioFlinger->readInput((uint32_t*) mInput,
		(uint32_t)mInputClientId,
		mRsmpInBuffer,
		mInputBytes,
		NULL);

		#else
		mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
		#endif
		mRsmpInIndex = 0;
		}
		if (mBytesRead < 0) {
		@@ -5038,7 +5078,15 @@ status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer*
		int channelCount;

		if (framesReady == 0) {
		#ifdef STE_AUDIO
		mBytesRead = mAudioFlinger->readInput((uint32_t*) mInput,
		(uint32_t)mInputClientId,
		mRsmpInBuffer,
		mInputBytes,
		NULL);
		#else
		mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
		#endif
		if (mBytesRead < 0) {
		LOGE("RecordThread::getNextBuffer() Error reading audio input");
		if (mActiveTrack->mState == TrackBase::ACTIVE) {
		@@ -5567,11 +5615,49 @@ status_t AudioFlinger::restoreOutput(int output)
		return NO_ERROR;
		}

		#ifdef STE_AUDIO
		uint32_t *AudioFlinger::addInputClient(uint32_t clientId)
		{
		Mutex::Autolock _l(mLock);

		uint32_t *pNewClient = new uint32_t;
		if (pNewClient) {
		*pNewClient = clientId;
		mInputClients.add(pNewClient);
		}

		return pNewClient;
		}

		status_t AudioFlinger::removeInputClient(uint32_t *pClientId)
		{
		status_t result = NO_ERROR;

		Mutex::Autolock _l(mLock);

		if (pClientId == NULL) {
		result = BAD_VALUE;
		} else if (mInputClients.remove(pClientId) < 0) {
		result = BAD_VALUE;
		} else {
		// the pointer was found in the vector and is non-NULL, so it must point to memory
		// allocated by addInputClient => free it.
		delete pClientId;
		}

		return result;
		}
		#endif
		int AudioFlinger::openInput(uint32_t *pDevices,
		uint32_t *pSamplingRate,
		uint32_t *pFormat,
		uint32_t *pChannels,
		#ifdef STE_AUDIO
		uint32_t acoustics,
		uint32_t *pInputClientId)
		#else
		uint32_t acoustics)
		#endif
		{
		status_t status;
		RecordThread *thread = NULL;
		@@ -5583,7 +5669,10 @@ int AudioFlinger::openInput(uint32_t *pDevices,
		uint32_t reqChannels = channels;
		audio_stream_in_t *inStream;
		audio_hw_device_t *inHwDev;

		#ifdef STE_AUDIO
		bool returnRecordThread = true;
		audio_input_clients inputClientId;
		#endif
		if (pDevices == NULL \|\| *pDevices == 0) {
		return 0;
		}
		@@ -5594,6 +5683,12 @@ int AudioFlinger::openInput(uint32_t *pDevices,
		if (inHwDev == NULL)
		return 0;

		#ifdef STE_AUDIO
		if (pInputClientId != NULL && *pInputClientId == AUDIO_INPUT_CLIENT_PLAYBACK) {
		returnRecordThread = false;
		}
		#endif

		status = inHwDev->open_input_stream(inHwDev, pDevices, (int )&format,
		&channels, &samplingRate,
		(audio_in_acoustics_t)acoustics,
		@@ -5620,6 +5715,7 @@ int AudioFlinger::openInput(uint32_t *pDevices,
		&inStream);
		}


		if (inStream != NULL) {
		AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);

		@@ -5633,13 +5729,22 @@ int AudioFlinger::openInput(uint32_t *pDevices,
		reqSamplingRate,
		reqChannels,
		id,
		#ifdef STE_AUDIO
		device,
		inputClientId);
		#else
		device);
		#endif
		mRecordThreads.add(id, thread);
		LOGV("openInput() created record thread: ID %d thread %p", id, thread);
		if (pSamplingRate) *pSamplingRate = reqSamplingRate;
		if (pFormat) *pFormat = format;
		if (pChannels) *pChannels = reqChannels;

		#ifdef STE_AUDIO
		if (pInputClientId != NULL) {
		*pInputClientId = inputClientId;
		}
		#endif
		input->stream->common.standby(&input->stream->common);

		// notify client processes of the new input creation
		@@ -5650,10 +5755,17 @@ int AudioFlinger::openInput(uint32_t *pDevices,
		return 0;
		}

		#ifdef STE_AUDIO
		status_t AudioFlinger::closeInput(int input, uint32_t *inputClientId)
		#else
		status_t AudioFlinger::closeInput(int input)
		#endif
		{
		// keep strong reference on the record thread so that
		// it is not destroyed while exit() is executed
		#ifdef STE_AUDIO
		audio_input_clients clientId = (audio_input_clients) *inputClientId;
		#endif
		sp <RecordThread> thread;
		{
		Mutex::Autolock _l(mLock);
		@@ -5675,6 +5787,7 @@ status_t AudioFlinger::closeInput(int input)
		delete in;

		return NO_ERROR;

		}

		status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
		@@ -6187,7 +6300,6 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId,
		return NO_ERROR;
		}


		// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
		sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
		const sp<AudioFlinger::Client>& client,