Merge "Prepare for transmitting audio through AudioSource." into jb-mr1-dev (86b1961a) · Commits · e / os / android_frameworks_av

include/media/stagefright/AudioSource.h

+14 −5

Original line number	Diff line number	Diff line
		@@ -31,11 +31,12 @@ namespace android {
		class AudioRecord;

		struct AudioSource : public MediaSource, public MediaBufferObserver {
		// Note that the "channels" parameter is _not_ the number of channels,
		// but a bitmask of audio_channels_t constants.
		// Note that the "channels" parameter _is_ the number of channels,
		// _not_ a bitmask of audio_channels_t constants.
		AudioSource(
		audio_source_t inputSource, uint32_t sampleRate,
		uint32_t channels = AUDIO_CHANNEL_IN_MONO);
		audio_source_t inputSource,
		uint32_t sampleRate,
		uint32_t channels = 1);

		status_t initCheck() const;

		@@ -49,9 +50,15 @@ struct AudioSource : public MediaSource, public MediaBufferObserver {
		virtual status_t read(
		MediaBuffer *buffer, const ReadOptions options = NULL);

		status_t dataCallbackTimestamp(const AudioRecord::Buffer& buffer, int64_t timeUs);
		status_t dataCallback(const AudioRecord::Buffer& buffer);
		virtual void signalBufferReturned(MediaBuffer *buffer);

		// If useLooperTime == true, buffers will carry absolute timestamps
		// as returned by ALooper::GetNowUs(), otherwise systemTime() is used
		// and buffers contain timestamps relative to start time.
		// The default is to _not_ use looper time.
		void setUseLooperTime(bool useLooperTime);

		protected:
		virtual ~AudioSource();

		@@ -87,6 +94,8 @@ private:

		List<MediaBuffer * > mBuffersReceived;

		bool mUseLooperTime;

		void trackMaxAmplitude(int16_t *data, int nSamples);

		// This is used to raise the volume from mute to the

media/libstagefright/AudioSource.cpp

+18 −5

Original line number	Diff line number	Diff line
		@@ -24,6 +24,7 @@
		#include <media/stagefright/MediaDefs.h>
		#include <media/stagefright/MetaData.h>
		#include <media/stagefright/foundation/ADebug.h>
		#include <media/stagefright/foundation/ALooper.h>
		#include <cutils/properties.h>
		#include <stdlib.h>

		@@ -33,7 +34,7 @@ static void AudioRecordCallbackFunction(int event, void user, void info) {
		AudioSource source = (AudioSource ) user;
		switch (event) {
		case AudioRecord::EVENT_MORE_DATA: {
		source->dataCallbackTimestamp(((AudioRecord::Buffer ) info), systemTime() / 1000);
		source->dataCallback(((AudioRecord::Buffer ) info));
		break;
		}
		case AudioRecord::EVENT_OVERRUN: {
		@@ -53,7 +54,8 @@ AudioSource::AudioSource(
		mSampleRate(sampleRate),
		mPrevSampleTimeUs(0),
		mNumFramesReceived(0),
		mNumClientOwnedBuffers(0) {
		mNumClientOwnedBuffers(0),
		mUseLooperTime(false) {

		ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount);
		CHECK(channelCount == 1 \|\| channelCount == 2);
		@@ -100,6 +102,12 @@ status_t AudioSource::initCheck() const {
		return mInitCheck;
		}

		void AudioSource::setUseLooperTime(bool useLooperTime) {
		CHECK(!mStarted);

		mUseLooperTime = useLooperTime;
		}

		status_t AudioSource::start(MetaData *params) {
		Mutex::Autolock autoLock(mLock);
		if (mStarted) {
		@@ -271,8 +279,10 @@ void AudioSource::signalBufferReturned(MediaBuffer *buffer) {
		return;
		}

		status_t AudioSource::dataCallbackTimestamp(
		const AudioRecord::Buffer& audioBuffer, int64_t timeUs) {
		status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
		int64_t timeUs =
		mUseLooperTime ? ALooper::GetNowUs() : (systemTime() / 1000ll);

		ALOGV("dataCallbackTimestamp: %lld us", timeUs);
		Mutex::Autolock autoLock(mLock);
		if (!mStarted) {
		@@ -290,12 +300,15 @@ status_t AudioSource::dataCallbackTimestamp(
		if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) {
		mInitialReadTimeUs = timeUs;
		// Initial delay
		if (mStartTimeUs > 0) {
		if (mUseLooperTime) {
		mStartTimeUs = timeUs;
		} else if (mStartTimeUs > 0) {
		mStartTimeUs = timeUs - mStartTimeUs;
		} else {
		// Assume latency is constant.
		mStartTimeUs += mRecord->latency() * 1000;
		}

		mPrevSampleTimeUs = mStartTimeUs;
		}

media/libstagefright/wifi-display/source/PlaybackSession.cpp

+27 −0

Original line number	Diff line number	Diff line
		@@ -32,6 +32,7 @@
		#include <media/stagefright/foundation/ADebug.h>
		#include <media/stagefright/foundation/AMessage.h>
		#include <media/stagefright/foundation/hexdump.h>
		#include <media/stagefright/AudioSource.h>
		#include <media/stagefright/DataSource.h>
		#include <media/stagefright/MediaDefs.h>
		#include <media/stagefright/MediaErrors.h>
		@@ -649,6 +650,32 @@ status_t WifiDisplaySource::PlaybackSession::setupPacketizer() {
		mTracks.add(index, new Track(converter));
		#endif

		#if 0
		sp<AudioSource> audioSource = new AudioSource(
		AUDIO_SOURCE_MIC,
		48000 /* sampleRate */,
		2 /* channelCount */); // XXX AUDIO_CHANNEL_IN_STEREO?

		CHECK_EQ((status_t)OK, audioSource->initCheck());

		audioSource->setUseLooperTime(true);

		index = mSerializer->addSource(audioSource);
		CHECK_GE(index, 0);

		sp<AMessage> audioFormat;
		err = convertMetaDataToMessage(audioSource->getFormat(), &audioFormat);
		CHECK_EQ(err, (status_t)OK);

		sp<AMessage> audioNotify = new AMessage(kWhatConverterNotify, id());
		audioNotify->setSize("trackIndex", index);

		converter = new Converter(audioNotify, mCodecLooper, audioFormat);
		looper()->registerHandler(converter);

		mTracks.add(index, new Track(converter));
		#endif

		return OK;
		}