const methods and comments (0bf65bdd) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioFlinger.cpp

+9 −9

Original line number	Diff line number	Diff line
		@@ -1924,7 +1924,7 @@ AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
		}

		// this method must always be called either with ThreadBase mLock held or inside the thread loop
		audio_stream_t* AudioFlinger::PlaybackThread::stream()
		audio_stream_t* AudioFlinger::PlaybackThread::stream() const
		{
		if (mOutput == NULL) {
		return NULL;
		@@ -1932,7 +1932,7 @@ audio_stream_t* AudioFlinger::PlaybackThread::stream()
		return &mOutput->stream->common;
		}

		uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
		uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
		{
		// A2DP output latency is not due only to buffering capacity. It also reflects encoding,
		// decoding and transfer time. So sleeping for half of the latency would likely cause
		@@ -2817,12 +2817,12 @@ status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>
		return NO_ERROR;
		}

		uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
		uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
		{
		return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
		}

		uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
		uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
		{
		return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
		}
		@@ -3166,7 +3166,7 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
		return reconfig;
		}

		uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
		uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
		{
		uint32_t time;
		if (audio_is_linear_pcm(mFormat)) {
		@@ -3177,7 +3177,7 @@ uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
		return time;
		}

		uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
		uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
		{
		uint32_t time;
		if (audio_is_linear_pcm(mFormat)) {
		@@ -3188,7 +3188,7 @@ uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
		return time;
		}

		uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
		uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
		{
		uint32_t time;
		if (audio_is_linear_pcm(mFormat)) {
		@@ -3351,7 +3351,7 @@ bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<Output
		return true;
		}

		uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
		uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
		{
		return (mWaitTimeMs * 1000) / 2;
		}
		@@ -5623,7 +5623,7 @@ AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
		}

		// this method must always be called either with ThreadBase mLock held or inside the thread loop
		audio_stream_t* AudioFlinger::RecordThread::stream()
		audio_stream_t* AudioFlinger::RecordThread::stream() const
		{
		if (mInput == NULL) {
		return NULL;

services/audioflinger/AudioFlinger.h

+17 −13

Original line number	Diff line number	Diff line
		@@ -472,7 +472,7 @@ private:
		audio_io_handle_t id() const { return mId;}
		bool standby() const { return mStandby; }
		uint32_t device() const { return mDevice; }
		virtual audio_stream_t* stream() = 0;
		virtual audio_stream_t* stream() const = 0;

		sp<EffectHandle> createEffect_l(
		const sp<AudioFlinger::Client>& client,
		@@ -918,7 +918,7 @@ public:

		AudioStreamOut* getOutput() const;
		AudioStreamOut* clearOutput();
		virtual audio_stream_t* stream();
		virtual audio_stream_t* stream() const;

		void suspend() { mSuspended++; }
		void restore() { if (mSuspended > 0) mSuspended--; }
		@@ -960,9 +960,13 @@ public:
		// Allocate a track name. Returns name >= 0 if successful, -1 on failure.
		virtual int getTrackName_l() = 0;
		virtual void deleteTrackName_l(int name) = 0;
		virtual uint32_t activeSleepTimeUs();
		virtual uint32_t idleSleepTimeUs() = 0;
		virtual uint32_t suspendSleepTimeUs() = 0;

		// Time to sleep between cycles when:
		virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED
		virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE
		virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us
		// No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
		// No sleep in standby mode; waits on a condition

		// Code snippets that are temporarily lifted up out of threadLoop() until the merge
		void checkSilentMode_l();
		@@ -1048,8 +1052,8 @@ public:
		virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
		virtual int getTrackName_l();
		virtual void deleteTrackName_l(int name);
		virtual uint32_t idleSleepTimeUs();
		virtual uint32_t suspendSleepTimeUs();
		virtual uint32_t idleSleepTimeUs() const;
		virtual uint32_t suspendSleepTimeUs() const;
		virtual void cacheParameters_l();

		// threadLoop snippets
		@@ -1073,9 +1077,9 @@ public:
		protected:
		virtual int getTrackName_l();
		virtual void deleteTrackName_l(int name);
		virtual uint32_t activeSleepTimeUs();
		virtual uint32_t idleSleepTimeUs();
		virtual uint32_t suspendSleepTimeUs();
		virtual uint32_t activeSleepTimeUs() const;
		virtual uint32_t idleSleepTimeUs() const;
		virtual uint32_t suspendSleepTimeUs() const;
		virtual void cacheParameters_l();

		// threadLoop snippets
		@@ -1110,9 +1114,9 @@ private:
		// Thread virtuals
		void addOutputTrack(MixerThread* thread);
		void removeOutputTrack(MixerThread* thread);
		uint32_t waitTimeMs() { return mWaitTimeMs; }
		uint32_t waitTimeMs() const { return mWaitTimeMs; }
		protected:
		virtual uint32_t activeSleepTimeUs();
		virtual uint32_t activeSleepTimeUs() const;

		private:
		bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
		@@ -1260,7 +1264,7 @@ private:
		status_t dump(int fd, const Vector<String16>& args);
		AudioStreamIn* getInput() const;
		AudioStreamIn* clearInput();
		virtual audio_stream_t* stream();
		virtual audio_stream_t* stream() const;

		// AudioBufferProvider interface
		virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);