diff options
Diffstat (limited to 'services/audioflinger/Threads.h')
| -rw-r--r-- | services/audioflinger/Threads.h | 37 |
1 files changed, 25 insertions, 12 deletions
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index a0b53cb..43e335d 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); @@ -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); @@ -275,6 +278,7 @@ protected: 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 @@ -358,7 +362,6 @@ public: void dump(int fd, const Vector<String16>& args); // Thread virtuals - virtual status_t readyToRun(); virtual bool threadLoop(); // RefBase @@ -425,7 +428,7 @@ public: IAudioFlinger::track_flags_t *flags, pid_t tid, int uid, - status_t *status); + status_t *status /*non-NULL*/); AudioStreamOut* getOutput() const; AudioStreamOut* clearOutput(); @@ -479,7 +482,6 @@ protected: 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 @@ -867,12 +869,12 @@ 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, @@ -883,7 +885,7 @@ public: 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, @@ -926,13 +928,13 @@ public: bool hasFastRecorder() const { return false; } private: - void clearSyncStartEvent(); + 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; @@ -945,11 +947,22 @@ private: 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() + + // 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 + size_t mRsmpInUnrel; // unreleased frames remaining from + // most recent getNextBuffer + // these are rolling counters that are never cleared + int32_t mRsmpInFront; // next available frame + int32_t mRsmpInRear; // last filled frame + 1 + size_t mRsmpInIndex; // FIXME legacy + + // client's requested configuration, which may differ from the HAL configuration 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. |