/* //device/include/server/AudioFlinger/AudioFlinger.h ** ** Copyright 2007, The Android Open Source Project ** ** Licensed under the Apache License, Version 2.0 (the "License"); ** you may not use this file except in compliance with the License. ** You may obtain a copy of the License at ** ** http://www.apache.org/licenses/LICENSE-2.0 ** ** Unless required by applicable law or agreed to in writing, software ** distributed under the License is distributed on an "AS IS" BASIS, ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. ** See the License for the specific language governing permissions and ** limitations under the License. */ #ifndef ANDROID_AUDIO_FLINGER_H #define ANDROID_AUDIO_FLINGER_H #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "AudioBufferProvider.h" namespace android { class audio_track_cblk_t; class AudioMixer; class AudioBuffer; // ---------------------------------------------------------------------------- #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) // ---------------------------------------------------------------------------- static const nsecs_t kStandbyTimeInNsecs = seconds(3); class AudioFlinger : public BnAudioFlinger, protected Thread, public IBinder::DeathRecipient { public: static void instantiate(); virtual status_t dump(int fd, const Vector& args); // Thread virtuals virtual bool threadLoop(); virtual status_t readyToRun(); virtual void onFirstRef(); // IAudioFlinger interface virtual sp createTrack( pid_t pid, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp& sharedBuffer, status_t *status); virtual uint32_t sampleRate() const; virtual int channelCount() const; virtual int format() const; virtual size_t frameCount() const; virtual size_t latency() const; virtual status_t setMasterVolume(float value); virtual status_t setMasterMute(bool muted); virtual float masterVolume() const; virtual bool masterMute() const; virtual status_t setStreamVolume(int stream, float value); virtual status_t setStreamMute(int stream, bool muted); virtual float streamVolume(int stream) const; virtual bool streamMute(int stream) const; virtual status_t setRouting(int mode, uint32_t routes, uint32_t mask); virtual uint32_t getRouting(int mode) const; virtual status_t setMode(int mode); virtual int getMode() const; virtual status_t setMicMute(bool state); virtual bool getMicMute() const; virtual bool isMusicActive() const; virtual status_t setParameter(const char* key, const char* value); virtual void registerClient(const sp& client); virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); virtual void wakeUp(); // IBinder::DeathRecipient virtual void binderDied(const wp& who); enum hardware_call_state { AUDIO_HW_IDLE = 0, AUDIO_HW_INIT, AUDIO_HW_OUTPUT_OPEN, AUDIO_HW_OUTPUT_CLOSE, AUDIO_HW_INPUT_OPEN, AUDIO_HW_INPUT_CLOSE, AUDIO_HW_STANDBY, AUDIO_HW_SET_MASTER_VOLUME, AUDIO_HW_GET_ROUTING, AUDIO_HW_SET_ROUTING, AUDIO_HW_GET_MODE, AUDIO_HW_SET_MODE, AUDIO_HW_GET_MIC_MUTE, AUDIO_HW_SET_MIC_MUTE, AUDIO_SET_VOICE_VOLUME, AUDIO_SET_PARAMETER, }; // record interface virtual sp openRecord( pid_t pid, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, status_t *status); virtual status_t onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); private: AudioFlinger(); virtual ~AudioFlinger(); void setOutput(AudioStreamOut* output); void doSetOutput(AudioStreamOut* output); size_t getOutputFrameCount(AudioStreamOut* output); #ifdef WITH_A2DP static bool streamDisablesA2dp(int streamType); inline bool isA2dpEnabled() const { return (mRequestedOutput == mA2dpOutput || (mOutput && mOutput == mA2dpOutput)); } void setA2dpEnabled(bool enable); #endif // Internal dump utilites. status_t dumpPermissionDenial(int fd, const Vector& args); status_t dumpClients(int fd, const Vector& args); status_t dumpTracks(int fd, const Vector& args); status_t dumpInternals(int fd, const Vector& args); // --- Client --- class Client : public RefBase { public: Client(const sp& audioFlinger, pid_t pid); virtual ~Client(); const sp& heap() const; pid_t pid() const { return mPid; } private: Client(const Client&); Client& operator = (const Client&); sp mAudioFlinger; sp mMemoryDealer; pid_t mPid; }; // --- Track --- class TrackHandle; class RecordHandle; class AudioRecordThread; // base for record and playback class TrackBase : public AudioBufferProvider, public RefBase { public: enum track_state { IDLE, TERMINATED, STOPPED, RESUMING, ACTIVE, PAUSING, PAUSED }; enum track_flags { STEPSERVER_FAILED = 0x01 // StepServer could not acquire cblk->lock mutex }; TrackBase( const sp& audioFlinger, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer); ~TrackBase(); virtual status_t start() = 0; virtual void stop() = 0; sp getCblk() const; protected: friend class AudioFlinger; friend class RecordHandle; friend class AudioRecordThread; TrackBase(const TrackBase&); TrackBase& operator = (const TrackBase&); virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0; virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); audio_track_cblk_t* cblk() const { return mCblk; } int type() const { return mStreamType; } int format() const { return mFormat; } int channelCount() const ; int sampleRate() const; void* getBuffer(uint32_t offset, uint32_t frames) const; int name() const { return mName; } bool isStopped() const { return mState == STOPPED; } bool isTerminated() const { return mState == TERMINATED; } bool step(); void reset(); sp mAudioFlinger; sp mClient; sp mCblkMemory; audio_track_cblk_t* mCblk; int mStreamType; void* mBuffer; void* mBufferEnd; uint32_t mFrameCount; int mName; // we don't really need a lock for these int mState; int mClientTid; uint8_t mFormat; uint8_t mFlags; }; // playback track class Track : public TrackBase { public: Track( const sp& audioFlinger, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer); ~Track(); void dump(char* buffer, size_t size); virtual status_t start(); virtual void stop(); void pause(); void flush(); void destroy(); void mute(bool); void setVolume(float left, float right); private: friend class AudioFlinger; friend class TrackHandle; Track(const Track&); Track& operator = (const Track&); virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); bool isMuted() const { return (mMute || mAudioFlinger->mStreamTypes[mStreamType].mute); } bool isPausing() const { return mState == PAUSING; } bool isPaused() const { return mState == PAUSED; } bool isReady() const; void setPaused() { mState = PAUSED; } void reset(); // we don't really need a lock for these float mVolume[2]; volatile bool mMute; // FILLED state is used for suppressing volume ramp at begin of playing enum {FS_FILLING, FS_FILLED, FS_ACTIVE}; mutable uint8_t mFillingUpStatus; int8_t mRetryCount; sp mSharedBuffer; bool mResetDone; }; // end of Track friend class AudioBuffer; class TrackHandle : public android::BnAudioTrack { public: TrackHandle(const sp& track); virtual ~TrackHandle(); virtual status_t start(); virtual void stop(); virtual void flush(); virtual void mute(bool); virtual void pause(); virtual void setVolume(float left, float right); virtual sp getCblk() const; virtual status_t onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); private: sp mTrack; }; struct stream_type_t { stream_type_t() : volume(1.0f), mute(false) { } float volume; bool mute; }; friend class Client; friend class Track; void removeClient(pid_t pid); status_t addTrack(const sp& track); void removeTrack(wp track, int name); void remove_track_l(wp track, int name); void destroyTrack(const sp& track); void addActiveTrack(const wp& track); void removeActiveTrack(const wp& track); int getTrackName(); void deleteTrackName(int name); AudioMixer* audioMixer() { return mAudioMixer; } // record track class RecordTrack : public TrackBase { public: RecordTrack( const sp& audioFlinger, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount); ~RecordTrack(); virtual status_t start(); virtual void stop(); bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } private: friend class AudioFlinger; friend class RecordHandle; friend class AudioRecordThread; RecordTrack(const Track&); RecordTrack& operator = (const Track&); virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); bool mOverflow; }; class RecordHandle : public android::BnAudioRecord { public: RecordHandle(const sp& recordTrack); virtual ~RecordHandle(); virtual status_t start(); virtual void stop(); virtual sp getCblk() const; virtual status_t onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); private: sp mRecordTrack; }; // record thread class AudioRecordThread : public Thread { public: AudioRecordThread(AudioHardwareInterface* audioHardware); virtual ~AudioRecordThread(); virtual bool threadLoop(); virtual status_t readyToRun() { return NO_ERROR; } virtual void onFirstRef() {} status_t start(RecordTrack* recordTrack); void stop(RecordTrack* recordTrack); void exit(); private: AudioRecordThread(); AudioHardwareInterface *mAudioHardware; sp mRecordTrack; Mutex mLock; Condition mWaitWorkCV; volatile bool mActive; status_t mStartStatus; }; friend class AudioRecordThread; status_t startRecord(RecordTrack* recordTrack); void stopRecord(RecordTrack* recordTrack); void notifyOutputChange_l(); mutable Mutex mHardwareLock; mutable Mutex mLock; mutable Condition mWaitWorkCV; DefaultKeyedVector< pid_t, wp > mClients; SortedVector< wp > mActiveTracks; SortedVector< sp > mTracks; float mMasterVolume; uint32_t mMasterRouting; bool mMasterMute; stream_type_t mStreamTypes[AudioTrack::NUM_STREAM_TYPES]; AudioMixer* mHardwareAudioMixer; AudioMixer* mA2dpAudioMixer; AudioMixer* mAudioMixer; AudioHardwareInterface* mAudioHardware; AudioHardwareInterface* mA2dpAudioInterface; AudioStreamOut* mHardwareOutput; AudioStreamOut* mA2dpOutput; AudioStreamOut* mOutput; AudioStreamOut* mRequestedOutput; sp mAudioRecordThread; uint32_t mSampleRate; size_t mFrameCount; int mChannelCount; int mFormat; int16_t* mMixBuffer; mutable int mHardwareStatus; nsecs_t mLastWriteTime; int mNumWrites; int mNumDelayedWrites; bool mStandby; bool mInWrite; int mA2dpDisableCount; bool mA2dpSuppressed; bool mMusicMuteSaved; SortedVector< wp > mNotificationClients; }; // ---------------------------------------------------------------------------- }; // namespace android #endif // ANDROID_AUDIO_FLINGER_H