Merge "Sample rate 0 means a route-dependent rate" into nyc-dev (db52e5e3) · Commits · e / os / android_frameworks_av

include/media/AudioRecord.h

+2 −1

Original line number	Diff line number	Diff line
		@@ -145,7 +145,7 @@ public:
		* Parameters:
		*
		* inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT).
		* sampleRate: Data sink sampling rate in Hz.
		* sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate.
		* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
		* 16 bits per sample).
		* channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true.
		@@ -258,6 +258,7 @@ public:
		bool stopped() const;

		/* Return the sink sample rate for this record track in Hz.
		* If specified as zero in constructor or set(), this will be the source sample rate.
		* Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock.
		*/
		uint32_t getSampleRate() const { return mSampleRate; }

include/media/AudioTrack.h

+9 −3

Original line number	Diff line number	Diff line
		@@ -167,7 +167,10 @@ public:
		*
		* streamType: Select the type of audio stream this track is attached to
		* (e.g. AUDIO_STREAM_MUSIC).
		* sampleRate: Data source sampling rate in Hz.
		* sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate.
		* A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set.
		* 0 will not work with current policy implementation for direct output
		* selection where an exact match is needed for sampling rate.
		* format: Audio format. For mixed tracks, any PCM format supported by server is OK.
		* For direct and offloaded tracks, the possible format(s) depends on the
		* output sink.
		@@ -395,11 +398,14 @@ public:
		status_t setAuxEffectSendLevel(float level);
		void getAuxEffectSendLevel(float* level) const;

		/* Set source sample rate for this track in Hz, mostly used for games' sound effects
		/* Set source sample rate for this track in Hz, mostly used for games' sound effects.
		* Zero is not permitted.
		*/
		status_t setSampleRate(uint32_t sampleRate);

		/* Return current source sample rate in Hz */
		/* Return current source sample rate in Hz.
		* If specified as zero in constructor or set(), this will be the sink sample rate.
		*/
		uint32_t getSampleRate() const;

		/* Return the original source sample rate in Hz. This corresponds to the sample rate

media/libmedia/AudioRecord.cpp

+62 −45

Original line number	Diff line number	Diff line
		@@ -191,10 +191,6 @@ status_t AudioRecord::set(
		mAttributes.source, mAttributes.flags, mAttributes.tags);
		}

		if (sampleRate == 0) {
		ALOGE("Invalid sample rate %u", sampleRate);
		return BAD_VALUE;
		}
		mSampleRate = sampleRate;

		// these below should probably come from the audioFlinger too...
		@@ -517,27 +513,71 @@ status_t AudioRecord::openRecord_l(const Modulo<uint32_t> &epoch, const String16
		return NO_INIT;
		}

		// Fast tracks must be at the primary _output_ [sic] sampling rate,
		// because there is currently no concept of a primary input sampling rate
		uint32_t afSampleRate = AudioSystem::getPrimaryOutputSamplingRate();
		if (afSampleRate == 0) {
		ALOGW("getPrimaryOutputSamplingRate failed");
		if (mDeviceCallback != 0 && mInput != AUDIO_IO_HANDLE_NONE) {
		AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mInput);
		}
		audio_io_handle_t input;

		status_t status;
		status = AudioSystem::getInputForAttr(&mAttributes, &input,
		(audio_session_t)mSessionId,
		// FIXME compare to AudioTrack
		IPCThreadState::self()->getCallingUid(),
		mSampleRate, mFormat, mChannelMask,
		mFlags, mSelectedDeviceId);

		if (status != NO_ERROR \|\| input == AUDIO_IO_HANDLE_NONE) {
		ALOGE("Could not get audio input for session %d, record source %d, sample rate %u, "
		"format %#x, channel mask %#x, flags %#x",
		mSessionId, mAttributes.source, mSampleRate, mFormat, mChannelMask, mFlags);
		return BAD_VALUE;
		}
		{
		// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
		// we must release it ourselves if anything goes wrong.

		#if 0
		size_t afFrameCount;
		status = AudioSystem::getFrameCount(input, &afFrameCount);
		if (status != NO_ERROR) {
		ALOGE("getFrameCount(input=%d) status %d", input, status);
		goto release;
		}
		#endif

		uint32_t afSampleRate;
		status = AudioSystem::getSamplingRate(input, &afSampleRate);
		if (status != NO_ERROR) {
		ALOGE("getSamplingRate(input=%d) status %d", input, status);
		goto release;
		}
		if (mSampleRate == 0) {
		mSampleRate = afSampleRate;
		}

		// Client can only express a preference for FAST. Server will perform additional tests.
		if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !((
		if (mFlags & AUDIO_INPUT_FLAG_FAST) {
		bool useCaseAllowed =
		// either of these use cases:
		// use case 1: callback transfer mode
		(mTransfer == TRANSFER_CALLBACK) \|\|
		// use case 2: obtain/release mode
		(mTransfer == TRANSFER_OBTAIN)) &&
		// matching sample rate
		(mSampleRate == afSampleRate))) {
		ALOGW("AUDIO_INPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, primary %u Hz",
		(mTransfer == TRANSFER_OBTAIN);
		// sample rates must also match
		bool fastAllowed = useCaseAllowed && (mSampleRate == afSampleRate);
		if (!fastAllowed) {
		ALOGW("AUDIO_INPUT_FLAG_FAST denied by client; transfer %d, "
		"track %u Hz, input %u Hz",
		mTransfer, mSampleRate, afSampleRate);
		// once denied, do not request again if IAudioRecord is re-created
		mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
		}
		}

		// The notification frame count is the period between callbacks, as suggested by the client
		// but moderated by the server. For record, the calculations are done entirely on server side.
		size_t notificationFrames = mNotificationFramesReq;
		size_t frameCount = mReqFrameCount;

		IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;

		@@ -549,35 +589,10 @@ status_t AudioRecord::openRecord_l(const Modulo<uint32_t> &epoch, const String16
		}
		}

		if (mDeviceCallback != 0 && mInput != AUDIO_IO_HANDLE_NONE) {
		AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mInput);
		}

		audio_io_handle_t input;
		status_t status = AudioSystem::getInputForAttr(&mAttributes, &input,
		(audio_session_t)mSessionId,
		IPCThreadState::self()->getCallingUid(),
		mSampleRate, mFormat, mChannelMask,
		mFlags, mSelectedDeviceId);

		if (status != NO_ERROR) {
		ALOGE("Could not get audio input for record source %d, sample rate %u, format %#x, "
		"channel mask %#x, session %d, flags %#x",
		mAttributes.source, mSampleRate, mFormat, mChannelMask, mSessionId, mFlags);
		return BAD_VALUE;
		}
		{
		// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
		// we must release it ourselves if anything goes wrong.

		size_t frameCount = mReqFrameCount;
		size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
		// but we will still need the original value also
		int originalSessionId = mSessionId;

		// The notification frame count is the period between callbacks, as suggested by the server.
		size_t notificationFrames = mNotificationFramesReq;

		sp<IMemory> iMem; // for cblk
		sp<IMemory> bufferMem;
		sp<IAudioRecord> record = audioFlinger->openRecord(input,
		@@ -660,11 +675,13 @@ status_t AudioRecord::openRecord_l(const Modulo<uint32_t> &epoch, const String16
		}
		}

		// Make sure that application is notified with sufficient margin before overrun
		if (notificationFrames == 0 \|\| notificationFrames > frameCount) {
		ALOGW("Received notificationFrames %zu for frameCount %zu", notificationFrames, frameCount);
		// Make sure that application is notified with sufficient margin before overrun.
		// The computation is done on server side.
		if (mNotificationFramesReq > 0 && notificationFrames != mNotificationFramesReq) {
		ALOGW("Server adjusted notificationFrames from %u to %zu for frameCount %zu",
		mNotificationFramesReq, notificationFrames, frameCount);
		}
		mNotificationFramesAct = notificationFrames;
		mNotificationFramesAct = (uint32_t) notificationFrames;

		// We retain a copy of the I/O handle, but don't own the reference
		mInput = input;

media/libmedia/AudioTrack.cpp

+21 −15

Original line number	Diff line number	Diff line
		@@ -1192,6 +1192,7 @@ status_t AudioTrack::createTrack_l()
		mSampleRate = mAfSampleRate;
		mOriginalSampleRate = mAfSampleRate;
		}

		// Client can only express a preference for FAST. Server will perform additional tests.
		if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
		bool useCaseAllowed =
		@@ -1215,13 +1216,6 @@ status_t AudioTrack::createTrack_l()
		}
		}

		// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
		// n = 1 fast track with single buffering; nBuffering is ignored
		// n = 2 fast track with double buffering
		// n = 2 normal track, (including those with sample rate conversion)
		// n >= 3 very high latency or very small notification interval (unused).
		const uint32_t nBuffering = 2;

		mNotificationFramesAct = mNotificationFramesReq;

		size_t frameCount = mReqFrameCount;
		@@ -1320,6 +1314,7 @@ status_t AudioTrack::createTrack_l()
		// AudioFlinger now owns the reference to the I/O handle,
		// so we are no longer responsible for releasing it.

		// FIXME compare to AudioRecord
		sp<IMemory> iMem = track->getCblk();
		if (iMem == 0) {
		ALOGE("Could not get control block");
		@@ -1383,14 +1378,25 @@ status_t AudioTrack::createTrack_l()
		//return NO_INIT;
		}
		}
		// Make sure that application is notified with sufficient margin before underrun

		// Make sure that application is notified with sufficient margin before underrun.
		// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
		// n = 1 fast track with single buffering; nBuffering is ignored
		// n = 2 fast track with double buffering
		// n = 2 normal track, (including those with sample rate conversion)
		// n >= 3 very high latency or very small notification interval (unused).
		// FIXME Move the computation from client side to server side,
		// and allow nBuffering to be larger than 1 for OpenSL ES, like it can be for Java.
		if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
		// Theoretically double-buffering is not required for fast tracks,
		// due to tighter scheduling. But in practice, to accommodate kernels with
		// scheduling jitter, and apps with computation jitter, we use double-buffering
		// for fast tracks just like normal streaming tracks.
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > frameCount / nBuffering) {
		mNotificationFramesAct = frameCount / nBuffering;
		size_t maxNotificationFrames = frameCount;
		if (!(trackFlags & IAudioFlinger::TRACK_FAST)) {
		const uint32_t nBuffering = 2;
		maxNotificationFrames /= nBuffering;
		}
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > maxNotificationFrames) {
		ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu",
		mNotificationFramesAct, maxNotificationFrames, frameCount);
		mNotificationFramesAct = (uint32_t) maxNotificationFrames;
		}
		}

services/audioflinger/AudioFlinger.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -564,6 +564,7 @@ private:
		PlaybackThread *srcThread,
		PlaybackThread *dstThread,
		bool reRegister);

		// return thread associated with primary hardware device, or NULL
		PlaybackThread *primaryPlaybackThread_l() const;
		audio_devices_t primaryOutputDevice_l() const;