diff options
Diffstat (limited to 'services/audioflinger/Threads.h')
| -rw-r--r-- | services/audioflinger/Threads.h | 106 |
1 files changed, 52 insertions, 54 deletions
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index a2fb874..fa3563c 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -36,6 +36,8 @@ public: audio_devices_t outDevice, audio_devices_t inDevice, type_t type); virtual ~ThreadBase(); + virtual status_t readyToRun(); + void dumpBase(int fd, const Vector<String16>& args); void dumpEffectChains(int fd, const Vector<String16>& args); @@ -63,7 +65,7 @@ public: class IoConfigEvent : public ConfigEvent { public: IoConfigEvent(int event, int param) : - ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} + ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(param) {} virtual ~IoConfigEvent() {} int event() const { return mEvent; } @@ -141,6 +143,7 @@ public: void sendIoConfigEvent_l(int event, int param = 0); void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); void processConfigEvents(); + void processConfigEvents_l(); // see note at declaration of mStandby, mOutDevice and mInDevice bool standby() const { return mStandby; } @@ -156,7 +159,7 @@ public: int sessionId, effect_descriptor_t *desc, int *enabled, - status_t *status); + status_t *status /*non-NULL*/); void disconnectEffect(const sp< EffectModule>& effect, EffectHandle *handle, bool unpinIfLast); @@ -198,13 +201,13 @@ public: // effect void removeEffect_l(const sp< EffectModule>& effect); // detach all tracks connected to an auxiliary effect - virtual void detachAuxEffect_l(int effectId) {} + virtual void detachAuxEffect_l(int effectId __unused) {} // returns either EFFECT_SESSION if effects on this audio session exist in one // chain, or TRACK_SESSION if tracks on this audio session exist, or both virtual uint32_t hasAudioSession(int sessionId) const = 0; // the value returned by default implementation is not important as the // strategy is only meaningful for PlaybackThread which implements this method - virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } + virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } // suspend or restore effect according to the type of effect passed. a NULL // type pointer means suspend all effects in the session @@ -267,14 +270,15 @@ protected: const sp<AudioFlinger> mAudioFlinger; - // updated by PlaybackThread::readOutputParameters() or - // RecordThread::readInputParameters() + // updated by PlaybackThread::readOutputParameters_l() or + // RecordThread::readInputParameters_l() uint32_t mSampleRate; size_t mFrameCount; // output HAL, direct output, record audio_channel_mask_t mChannelMask; uint32_t mChannelCount; size_t mFrameSize; audio_format_t mFormat; + size_t mBufferSize; // HAL buffer size for read() or write() // Parameter sequence by client: binder thread calling setParameters(): // 1. Lock mLock @@ -303,12 +307,12 @@ protected: Vector<ConfigEvent *> mConfigEvents; // These fields are written and read by thread itself without lock or barrier, - // and read by other threads without lock or barrier via standby() , outDevice() + // and read by other threads without lock or barrier via standby(), outDevice() // and inDevice(). // Because of the absence of a lock or barrier, any other thread that reads // these fields must use the information in isolation, or be prepared to deal // with possibility that it might be inconsistent with other information. - bool mStandby; // Whether thread is currently in standby. + bool mStandby; // Whether thread is currently in standby. audio_devices_t mOutDevice; // output device audio_devices_t mInDevice; // input device audio_source_t mAudioSource; // (see audio.h, audio_source_t) @@ -358,7 +362,6 @@ public: void dump(int fd, const Vector<String16>& args); // Thread virtuals - virtual status_t readyToRun(); virtual bool threadLoop(); // RefBase @@ -391,7 +394,7 @@ protected: virtual bool waitingAsyncCallback(); virtual bool waitingAsyncCallback_l(); virtual bool shouldStandby_l(); - + virtual void onAddNewTrack_l(); // ThreadBase virtuals virtual void preExit(); @@ -419,13 +422,13 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - size_t frameCount, + size_t *pFrameCount, const sp<IMemory>& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t *flags, pid_t tid, int uid, - status_t *status); + status_t *status /*non-NULL*/); AudioStreamOut* getOutput() const; AudioStreamOut* clearOutput(); @@ -475,11 +478,10 @@ public: status_t getTimestamp_l(AudioTimestamp& timestamp); protected: - // updated by readOutputParameters() + // updated by readOutputParameters_l() size_t mNormalFrameCount; // normal mixer and effects int16_t* mMixBuffer; // frame size aligned mix buffer - int8_t* mAllocMixBuffer; // mixer buffer allocation address // suspend count, > 0 means suspended. While suspended, the thread continues to pull from // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle @@ -539,7 +541,7 @@ private: void removeTrack_l(const sp<Track>& track); void broadcast_l(); - void readOutputParameters(); + void readOutputParameters_l(); virtual void dumpInternals(int fd, const Vector<String16>& args); void dumpTracks(int fd, const Vector<String16>& args); @@ -623,13 +625,12 @@ private: sp<NBLog::Writer> mFastMixerNBLogWriter; public: virtual bool hasFastMixer() const = 0; - virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const + virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const { FastTrackUnderruns dummy; return dummy; } protected: // accessed by both binder threads and within threadLoop(), lock on mutex needed unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available - virtual void flushOutput_l(); private: // timestamp latch: @@ -748,11 +749,11 @@ protected: // threadLoop snippets virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); virtual void threadLoop_exit(); - virtual void flushOutput_l(); virtual bool waitingAsyncCallback(); virtual bool waitingAsyncCallback_l(); virtual bool shouldStandby_l(); + virtual void onAddNewTrack_l(); private: void flushHw_l(); @@ -838,17 +839,28 @@ public: // record thread -class RecordThread : public ThreadBase, public AudioBufferProvider - // derives from AudioBufferProvider interface for use by resampler +class RecordThread : public ThreadBase { public: + class RecordTrack; + class ResamplerBufferProvider : public AudioBufferProvider + // derives from AudioBufferProvider interface for use by resampler + { + public: + ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { } + virtual ~ResamplerBufferProvider() { } + // AudioBufferProvider interface + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); + virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); + private: + RecordTrack * const mRecordTrack; + }; + #include "RecordTracks.h" RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, - uint32_t sampleRate, - audio_channel_mask_t channelMask, audio_io_handle_t id, audio_devices_t outDevice, audio_devices_t inDevice @@ -867,23 +879,23 @@ public: // Thread virtuals virtual bool threadLoop(); - virtual status_t readyToRun(); // RefBase virtual void onFirstRef(); virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } + sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - size_t frameCount, + size_t *pFrameCount, int sessionId, int uid, IAudioFlinger::track_flags_t *flags, pid_t tid, - status_t *status); + status_t *status /*non-NULL*/); status_t start(RecordTrack* recordTrack, AudioSystem::sync_event_t event, @@ -897,15 +909,12 @@ public: AudioStreamIn* clearInput(); virtual audio_stream_t* stream() const; - // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); - virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); virtual bool checkForNewParameters_l(); virtual String8 getParameters(const String8& keys); virtual void audioConfigChanged_l(int event, int param = 0); - void readInputParameters(); - virtual unsigned int getInputFramesLost(); + void readInputParameters_l(); + virtual uint32_t getInputFramesLost(); virtual status_t addEffectChain_l(const sp<EffectChain>& chain); virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); @@ -920,44 +929,33 @@ public: virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; static void syncStartEventCallback(const wp<SyncEvent>& event); - void handleSyncStartEvent(const sp<SyncEvent>& event); virtual size_t frameCount() const { return mFrameCount; } bool hasFastRecorder() const { return false; } private: - void clearSyncStartEvent(); - // Enter standby if not already in standby, and set mStandby flag - void standby(); + void standbyIfNotAlreadyInStandby(); // Call the HAL standby method unconditionally, and don't change mStandby flag - void inputStandBy(); + void inputStandBy(); AudioStreamIn *mInput; SortedVector < sp<RecordTrack> > mTracks; - // mActiveTrack has dual roles: it indicates the current active track, and + // mActiveTracks has dual roles: it indicates the current active track(s), and // is used together with mStartStopCond to indicate start()/stop() progress - sp<RecordTrack> mActiveTrack; + SortedVector< sp<RecordTrack> > mActiveTracks; + // generation counter for mActiveTracks + int mActiveTracksGen; Condition mStartStopCond; - // updated by RecordThread::readInputParameters() - AudioResampler *mResampler; - // interleaved stereo pairs of fixed-point signed Q19.12 - int32_t *mRsmpOutBuffer; - int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount] - size_t mRsmpInIndex; - size_t mBufferSize; // stream buffer size for read() - const uint32_t mReqChannelCount; - const uint32_t mReqSampleRate; - ssize_t mBytesRead; - // sync event triggering actual audio capture. Frames read before this event will - // be dropped and therefore not read by the application. - sp<SyncEvent> mSyncStartEvent; - // number of captured frames to drop after the start sync event has been received. - // when < 0, maximum frames to drop before starting capture even if sync event is - // not received - ssize_t mFramestoDrop; + // resampler converts input at HAL Hz to output at AudioRecord client Hz + int16_t *mRsmpInBuffer; // see new[] for details on the size + size_t mRsmpInFrames; // size of resampler input in frames + size_t mRsmpInFramesP2;// size rounded up to a power-of-2 + + // rolling index that is never cleared + int32_t mRsmpInRear; // last filled frame + 1 // For dumpsys const sp<NBAIO_Sink> mTeeSink; |