| /* |
| ** |
| ** 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 <stdint.h> |
| #include <sys/types.h> |
| #include <limits.h> |
| |
| #include <common_time/cc_helper.h> |
| |
| #include <media/IAudioFlinger.h> |
| #include <media/IAudioFlingerClient.h> |
| #include <media/IAudioTrack.h> |
| #include <media/IAudioRecord.h> |
| #include <media/AudioSystem.h> |
| #include <media/AudioTrack.h> |
| |
| #include <utils/Atomic.h> |
| #include <utils/Errors.h> |
| #include <utils/threads.h> |
| #include <utils/SortedVector.h> |
| #include <utils/TypeHelpers.h> |
| #include <utils/Vector.h> |
| |
| #include <binder/BinderService.h> |
| #include <binder/MemoryDealer.h> |
| |
| #include <system/audio.h> |
| #include <hardware/audio.h> |
| |
| #include "AudioBufferProvider.h" |
| |
| #include <powermanager/IPowerManager.h> |
| |
| namespace android { |
| |
| class audio_track_cblk_t; |
| class effect_param_cblk_t; |
| class AudioMixer; |
| class AudioBuffer; |
| class AudioResampler; |
| |
| // ---------------------------------------------------------------------------- |
| |
| // AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. |
| // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. |
| // Adding full support for > 2 channel capture or playback would require more than simply changing |
| // this #define. There is an independent hard-coded upper limit in AudioMixer; |
| // removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. |
| // The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. |
| // Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. |
| #define FCC_2 2 // FCC_2 = Fixed Channel Count 2 |
| |
| static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); |
| |
| class AudioFlinger : |
| public BinderService<AudioFlinger>, |
| public BnAudioFlinger |
| { |
| friend class BinderService<AudioFlinger>; // for AudioFlinger() |
| public: |
| static const char* getServiceName() { return "media.audio_flinger"; } |
| |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| |
| // IAudioFlinger interface, in binder opcode order |
| virtual sp<IAudioTrack> createTrack( |
| pid_t pid, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| uint32_t flags, |
| const sp<IMemory>& sharedBuffer, |
| audio_io_handle_t output, |
| bool isTimed, |
| int *sessionId, |
| status_t *status); |
| |
| virtual sp<IAudioRecord> openRecord( |
| pid_t pid, |
| audio_io_handle_t input, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| uint32_t flags, |
| int *sessionId, |
| status_t *status); |
| |
| virtual uint32_t sampleRate(audio_io_handle_t output) const; |
| virtual int channelCount(audio_io_handle_t output) const; |
| virtual audio_format_t format(audio_io_handle_t output) const; |
| virtual size_t frameCount(audio_io_handle_t output) const; |
| virtual uint32_t latency(audio_io_handle_t output) const; |
| |
| virtual status_t setMasterVolume(float value); |
| virtual status_t setMasterMute(bool muted); |
| |
| virtual float masterVolume() const; |
| virtual float masterVolumeSW() const; |
| virtual bool masterMute() const; |
| |
| virtual status_t setStreamVolume(audio_stream_type_t stream, float value, |
| audio_io_handle_t output); |
| virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); |
| |
| virtual float streamVolume(audio_stream_type_t stream, |
| audio_io_handle_t output) const; |
| virtual bool streamMute(audio_stream_type_t stream) const; |
| |
| virtual status_t setMode(audio_mode_t mode); |
| |
| virtual status_t setMicMute(bool state); |
| virtual bool getMicMute() const; |
| |
| virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); |
| virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; |
| |
| virtual void registerClient(const sp<IAudioFlingerClient>& client); |
| |
| virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; |
| |
| virtual audio_io_handle_t openOutput(uint32_t *pDevices, |
| uint32_t *pSamplingRate, |
| audio_format_t *pFormat, |
| uint32_t *pChannels, |
| uint32_t *pLatencyMs, |
| audio_policy_output_flags_t flags); |
| |
| virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, |
| audio_io_handle_t output2); |
| |
| virtual status_t closeOutput(audio_io_handle_t output); |
| |
| virtual status_t suspendOutput(audio_io_handle_t output); |
| |
| virtual status_t restoreOutput(audio_io_handle_t output); |
| |
| virtual audio_io_handle_t openInput(uint32_t *pDevices, |
| uint32_t *pSamplingRate, |
| audio_format_t *pFormat, |
| uint32_t *pChannels, |
| audio_in_acoustics_t acoustics); |
| |
| virtual status_t closeInput(audio_io_handle_t input); |
| |
| virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); |
| |
| virtual status_t setVoiceVolume(float volume); |
| |
| virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, |
| audio_io_handle_t output) const; |
| |
| virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; |
| |
| virtual int newAudioSessionId(); |
| |
| virtual void acquireAudioSessionId(int audioSession); |
| |
| virtual void releaseAudioSessionId(int audioSession); |
| |
| virtual status_t queryNumberEffects(uint32_t *numEffects) const; |
| |
| virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; |
| |
| virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, |
| effect_descriptor_t *descriptor) const; |
| |
| virtual sp<IEffect> createEffect(pid_t pid, |
| effect_descriptor_t *pDesc, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority, |
| audio_io_handle_t io, |
| int sessionId, |
| status_t *status, |
| int *id, |
| int *enabled); |
| |
| virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, |
| audio_io_handle_t dstOutput); |
| |
| virtual status_t onTransact( |
| uint32_t code, |
| const Parcel& data, |
| Parcel* reply, |
| uint32_t flags); |
| |
| // end of IAudioFlinger interface |
| |
| private: |
| audio_mode_t getMode() const { return mMode; } |
| |
| bool btNrecIsOff() const { return mBtNrecIsOff; } |
| |
| AudioFlinger(); |
| virtual ~AudioFlinger(); |
| |
| // call in any IAudioFlinger method that accesses mPrimaryHardwareDev |
| status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } |
| |
| // RefBase |
| virtual void onFirstRef(); |
| |
| audio_hw_device_t* findSuitableHwDev_l(uint32_t devices); |
| void purgeStaleEffects_l(); |
| |
| // standby delay for MIXER and DUPLICATING playback threads is read from property |
| // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs |
| static nsecs_t mStandbyTimeInNsecs; |
| |
| // Internal dump utilites. |
| status_t dumpPermissionDenial(int fd, const Vector<String16>& args); |
| status_t dumpClients(int fd, const Vector<String16>& args); |
| status_t dumpInternals(int fd, const Vector<String16>& args); |
| |
| // --- Client --- |
| class Client : public RefBase { |
| public: |
| Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); |
| virtual ~Client(); |
| sp<MemoryDealer> heap() const; |
| pid_t pid() const { return mPid; } |
| sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } |
| |
| bool reserveTimedTrack(); |
| void releaseTimedTrack(); |
| |
| private: |
| Client(const Client&); |
| Client& operator = (const Client&); |
| const sp<AudioFlinger> mAudioFlinger; |
| const sp<MemoryDealer> mMemoryDealer; |
| const pid_t mPid; |
| |
| Mutex mTimedTrackLock; |
| int mTimedTrackCount; |
| }; |
| |
| // --- Notification Client --- |
| class NotificationClient : public IBinder::DeathRecipient { |
| public: |
| NotificationClient(const sp<AudioFlinger>& audioFlinger, |
| const sp<IAudioFlingerClient>& client, |
| pid_t pid); |
| virtual ~NotificationClient(); |
| |
| sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } |
| |
| // IBinder::DeathRecipient |
| virtual void binderDied(const wp<IBinder>& who); |
| |
| private: |
| NotificationClient(const NotificationClient&); |
| NotificationClient& operator = (const NotificationClient&); |
| |
| const sp<AudioFlinger> mAudioFlinger; |
| const pid_t mPid; |
| const sp<IAudioFlingerClient> mAudioFlingerClient; |
| }; |
| |
| class TrackHandle; |
| class RecordHandle; |
| class RecordThread; |
| class PlaybackThread; |
| class MixerThread; |
| class DirectOutputThread; |
| class DuplicatingThread; |
| class Track; |
| class RecordTrack; |
| class EffectModule; |
| class EffectHandle; |
| class EffectChain; |
| struct AudioStreamOut; |
| struct AudioStreamIn; |
| |
| class ThreadBase : public Thread { |
| public: |
| |
| enum type_t { |
| MIXER, // Thread class is MixerThread |
| DIRECT, // Thread class is DirectOutputThread |
| DUPLICATING, // Thread class is DuplicatingThread |
| RECORD // Thread class is RecordThread |
| }; |
| |
| ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); |
| virtual ~ThreadBase(); |
| |
| status_t dumpBase(int fd, const Vector<String16>& args); |
| status_t dumpEffectChains(int fd, const Vector<String16>& args); |
| |
| void clearPowerManager(); |
| |
| // base for record and playback |
| class TrackBase : public AudioBufferProvider, public RefBase { |
| |
| public: |
| enum track_state { |
| IDLE, |
| TERMINATED, |
| // These are order-sensitive; do not change order without reviewing the impact. |
| // In particular there are assumptions about > STOPPED. |
| STOPPED, |
| RESUMING, |
| ACTIVE, |
| PAUSING, |
| PAUSED |
| }; |
| |
| TrackBase(ThreadBase *thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId); |
| virtual ~TrackBase(); |
| |
| virtual status_t start(pid_t tid) = 0; |
| virtual void stop() = 0; |
| sp<IMemory> getCblk() const { return mCblkMemory; } |
| audio_track_cblk_t* cblk() const { return mCblk; } |
| int sessionId() const { return mSessionId; } |
| |
| protected: |
| TrackBase(const TrackBase&); |
| TrackBase& operator = (const TrackBase&); |
| |
| // AudioBufferProvider interface |
| virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; |
| virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); |
| |
| audio_format_t format() const { |
| return mFormat; |
| } |
| |
| int channelCount() const { return mChannelCount; } |
| |
| uint32_t channelMask() const { return mChannelMask; } |
| |
| int sampleRate() const; // FIXME inline after cblk sr moved |
| |
| void* getBuffer(uint32_t offset, uint32_t frames) const; |
| |
| bool isStopped() const { |
| return mState == STOPPED; |
| } |
| |
| bool isTerminated() const { |
| return mState == TERMINATED; |
| } |
| |
| bool step(); |
| void reset(); |
| |
| const wp<ThreadBase> mThread; |
| /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const |
| sp<IMemory> mCblkMemory; |
| audio_track_cblk_t* mCblk; |
| void* mBuffer; |
| void* mBufferEnd; |
| uint32_t mFrameCount; |
| // we don't really need a lock for these |
| track_state mState; |
| const audio_format_t mFormat; |
| bool mStepServerFailed; |
| const int mSessionId; |
| uint8_t mChannelCount; |
| uint32_t mChannelMask; |
| }; |
| |
| class ConfigEvent { |
| public: |
| ConfigEvent() : mEvent(0), mParam(0) {} |
| |
| int mEvent; |
| int mParam; |
| }; |
| |
| class PMDeathRecipient : public IBinder::DeathRecipient { |
| public: |
| PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} |
| virtual ~PMDeathRecipient() {} |
| |
| // IBinder::DeathRecipient |
| virtual void binderDied(const wp<IBinder>& who); |
| |
| private: |
| PMDeathRecipient(const PMDeathRecipient&); |
| PMDeathRecipient& operator = (const PMDeathRecipient&); |
| |
| wp<ThreadBase> mThread; |
| }; |
| |
| virtual status_t initCheck() const = 0; |
| type_t type() const { return mType; } |
| uint32_t sampleRate() const { return mSampleRate; } |
| int channelCount() const { return mChannelCount; } |
| audio_format_t format() const { return mFormat; } |
| size_t frameCount() const { return mFrameCount; } |
| void wakeUp() { mWaitWorkCV.broadcast(); } |
| // Should be "virtual status_t requestExitAndWait()" and override same |
| // method in Thread, but Thread::requestExitAndWait() is not yet virtual. |
| void exit(); |
| virtual bool checkForNewParameters_l() = 0; |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys) = 0; |
| virtual void audioConfigChanged_l(int event, int param = 0) = 0; |
| void sendConfigEvent(int event, int param = 0); |
| void sendConfigEvent_l(int event, int param = 0); |
| void processConfigEvents(); |
| 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; |
| |
| sp<EffectHandle> createEffect_l( |
| const sp<AudioFlinger::Client>& client, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority, |
| int sessionId, |
| effect_descriptor_t *desc, |
| int *enabled, |
| status_t *status); |
| void disconnectEffect(const sp< EffectModule>& effect, |
| const wp<EffectHandle>& handle, |
| bool unpinIfLast); |
| |
| // return values for hasAudioSession (bit field) |
| enum effect_state { |
| EFFECT_SESSION = 0x1, // the audio session corresponds to at least one |
| // effect |
| TRACK_SESSION = 0x2 // the audio session corresponds to at least one |
| // track |
| }; |
| |
| // get effect chain corresponding to session Id. |
| sp<EffectChain> getEffectChain(int sessionId); |
| // same as getEffectChain() but must be called with ThreadBase mutex locked |
| sp<EffectChain> getEffectChain_l(int sessionId); |
| // add an effect chain to the chain list (mEffectChains) |
| virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; |
| // remove an effect chain from the chain list (mEffectChains) |
| virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; |
| // lock all effect chains Mutexes. Must be called before releasing the |
| // ThreadBase mutex before processing the mixer and effects. This guarantees the |
| // integrity of the chains during the process. |
| // Also sets the parameter 'effectChains' to current value of mEffectChains. |
| void lockEffectChains_l(Vector<sp <EffectChain> >& effectChains); |
| // unlock effect chains after process |
| void unlockEffectChains(const Vector<sp<EffectChain> >& effectChains); |
| // set audio mode to all effect chains |
| void setMode(audio_mode_t mode); |
| // get effect module with corresponding ID on specified audio session |
| sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); |
| // add and effect module. Also creates the effect chain is none exists for |
| // the effects audio session |
| status_t addEffect_l(const sp< EffectModule>& effect); |
| // remove and effect module. Also removes the effect chain is this was the last |
| // effect |
| void removeEffect_l(const sp< EffectModule>& effect); |
| // detach all tracks connected to an auxiliary effect |
| virtual void detachAuxEffect_l(int effectId) {} |
| // 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) = 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; } |
| |
| // suspend or restore effect according to the type of effect passed. a NULL |
| // type pointer means suspend all effects in the session |
| void setEffectSuspended(const effect_uuid_t *type, |
| bool suspend, |
| int sessionId = AUDIO_SESSION_OUTPUT_MIX); |
| // check if some effects must be suspended/restored when an effect is enabled |
| // or disabled |
| void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, |
| bool enabled, |
| int sessionId = AUDIO_SESSION_OUTPUT_MIX); |
| void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, |
| bool enabled, |
| int sessionId = AUDIO_SESSION_OUTPUT_MIX); |
| mutable Mutex mLock; |
| |
| protected: |
| |
| // entry describing an effect being suspended in mSuspendedSessions keyed vector |
| class SuspendedSessionDesc : public RefBase { |
| public: |
| SuspendedSessionDesc() : mRefCount(0) {} |
| |
| int mRefCount; // number of active suspend requests |
| effect_uuid_t mType; // effect type UUID |
| }; |
| |
| void acquireWakeLock(); |
| void acquireWakeLock_l(); |
| void releaseWakeLock(); |
| void releaseWakeLock_l(); |
| void setEffectSuspended_l(const effect_uuid_t *type, |
| bool suspend, |
| int sessionId = AUDIO_SESSION_OUTPUT_MIX); |
| // updated mSuspendedSessions when an effect suspended or restored |
| void updateSuspendedSessions_l(const effect_uuid_t *type, |
| bool suspend, |
| int sessionId); |
| // check if some effects must be suspended when an effect chain is added |
| void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); |
| |
| friend class AudioFlinger; // for mEffectChains |
| |
| const type_t mType; |
| |
| // Used by parameters, config events, addTrack_l, exit |
| Condition mWaitWorkCV; |
| |
| const sp<AudioFlinger> mAudioFlinger; |
| uint32_t mSampleRate; |
| size_t mFrameCount; |
| uint32_t mChannelMask; |
| uint16_t mChannelCount; |
| size_t mFrameSize; |
| audio_format_t mFormat; |
| |
| // Parameter sequence by client: binder thread calling setParameters(): |
| // 1. Lock mLock |
| // 2. Append to mNewParameters |
| // 3. mWaitWorkCV.signal |
| // 4. mParamCond.waitRelative with timeout |
| // 5. read mParamStatus |
| // 6. mWaitWorkCV.signal |
| // 7. Unlock |
| // |
| // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): |
| // 1. Lock mLock |
| // 2. If there is an entry in mNewParameters proceed ... |
| // 2. Read first entry in mNewParameters |
| // 3. Process |
| // 4. Remove first entry from mNewParameters |
| // 5. Set mParamStatus |
| // 6. mParamCond.signal |
| // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) |
| // 8. Unlock |
| Condition mParamCond; |
| Vector<String8> mNewParameters; |
| status_t mParamStatus; |
| |
| Vector<ConfigEvent> mConfigEvents; |
| bool mStandby; |
| const audio_io_handle_t mId; |
| Vector< sp<EffectChain> > mEffectChains; |
| uint32_t mDevice; // output device for PlaybackThread |
| // input + output devices for RecordThread |
| static const int kNameLength = 16; // prctl(PR_SET_NAME) limit |
| char mName[kNameLength]; |
| sp<IPowerManager> mPowerManager; |
| sp<IBinder> mWakeLockToken; |
| const sp<PMDeathRecipient> mDeathRecipient; |
| // list of suspended effects per session and per type. The first vector is |
| // keyed by session ID, the second by type UUID timeLow field |
| KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; |
| }; |
| |
| struct stream_type_t { |
| stream_type_t() |
| : volume(1.0f), |
| mute(false), |
| valid(true) |
| { |
| } |
| float volume; |
| bool mute; |
| bool valid; |
| }; |
| |
| // --- PlaybackThread --- |
| class PlaybackThread : public ThreadBase { |
| public: |
| |
| enum mixer_state { |
| MIXER_IDLE, // no active tracks |
| MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready |
| MIXER_TRACKS_READY // at least one active track, and at least one track has data |
| // standby mode does not have an enum value |
| // suspend by audio policy manager is orthogonal to mixer state |
| }; |
| |
| // playback track |
| class Track : public TrackBase { |
| public: |
| Track( PlaybackThread *thread, |
| const sp<Client>& client, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId); |
| virtual ~Track(); |
| |
| void dump(char* buffer, size_t size); |
| virtual status_t start(pid_t tid); |
| virtual void stop(); |
| void pause(); |
| |
| void flush(); |
| void destroy(); |
| void mute(bool); |
| int name() const { |
| return mName; |
| } |
| |
| audio_stream_type_t streamType() const { |
| return mStreamType; |
| } |
| status_t attachAuxEffect(int EffectId); |
| void setAuxBuffer(int EffectId, int32_t *buffer); |
| int32_t *auxBuffer() const { return mAuxBuffer; } |
| void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } |
| int16_t *mainBuffer() const { return mMainBuffer; } |
| int auxEffectId() const { return mAuxEffectId; } |
| |
| protected: |
| // for numerous |
| friend class PlaybackThread; |
| friend class MixerThread; |
| friend class DirectOutputThread; |
| |
| Track(const Track&); |
| Track& operator = (const Track&); |
| |
| // AudioBufferProvider interface |
| virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); |
| // releaseBuffer() not overridden |
| |
| virtual uint32_t framesReady() const; |
| |
| bool isMuted() const { return mMute; } |
| bool isPausing() const { |
| return mState == PAUSING; |
| } |
| bool isPaused() const { |
| return mState == PAUSED; |
| } |
| bool isReady() const; |
| void setPaused() { mState = PAUSED; } |
| void reset(); |
| |
| bool isOutputTrack() const { |
| return (mStreamType == AUDIO_STREAM_CNT); |
| } |
| |
| public: |
| virtual bool isTimedTrack() const { return false; } |
| protected: |
| |
| // we don't really need a lock for these |
| 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<IMemory> mSharedBuffer; |
| bool mResetDone; |
| audio_stream_type_t mStreamType; |
| int mName; |
| int16_t *mMainBuffer; |
| int32_t *mAuxBuffer; |
| int mAuxEffectId; |
| bool mHasVolumeController; |
| }; // end of Track |
| |
| class TimedTrack : public Track { |
| public: |
| static sp<TimedTrack> create(PlaybackThread *thread, |
| const sp<Client>& client, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId); |
| ~TimedTrack(); |
| |
| class TimedBuffer { |
| public: |
| TimedBuffer(); |
| TimedBuffer(const sp<IMemory>& buffer, int64_t pts); |
| const sp<IMemory>& buffer() const { return mBuffer; } |
| int64_t pts() const { return mPTS; } |
| int position() const { return mPosition; } |
| void setPosition(int pos) { mPosition = pos; } |
| private: |
| sp<IMemory> mBuffer; |
| int64_t mPTS; |
| int mPosition; |
| }; |
| |
| virtual bool isTimedTrack() const { return true; } |
| |
| virtual uint32_t framesReady() const; |
| |
| // AudioBufferProvider interface |
| virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); |
| virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); |
| |
| void timedYieldSamples(AudioBufferProvider::Buffer* buffer); |
| void timedYieldSilence(uint32_t numFrames, |
| AudioBufferProvider::Buffer* buffer); |
| |
| status_t allocateTimedBuffer(size_t size, |
| sp<IMemory>* buffer); |
| status_t queueTimedBuffer(const sp<IMemory>& buffer, |
| int64_t pts); |
| status_t setMediaTimeTransform(const LinearTransform& xform, |
| TimedAudioTrack::TargetTimeline target); |
| void trimTimedBufferQueue_l(); |
| |
| private: |
| TimedTrack(PlaybackThread *thread, |
| const sp<Client>& client, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId); |
| |
| uint64_t mLocalTimeFreq; |
| LinearTransform mLocalTimeToSampleTransform; |
| sp<MemoryDealer> mTimedMemoryDealer; |
| Vector<TimedBuffer> mTimedBufferQueue; |
| uint8_t* mTimedSilenceBuffer; |
| uint32_t mTimedSilenceBufferSize; |
| mutable Mutex mTimedBufferQueueLock; |
| bool mTimedAudioOutputOnTime; |
| CCHelper mCCHelper; |
| |
| Mutex mMediaTimeTransformLock; |
| LinearTransform mMediaTimeTransform; |
| bool mMediaTimeTransformValid; |
| TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; |
| }; |
| |
| |
| // playback track |
| class OutputTrack : public Track { |
| public: |
| |
| class Buffer: public AudioBufferProvider::Buffer { |
| public: |
| int16_t *mBuffer; |
| }; |
| |
| OutputTrack(PlaybackThread *thread, |
| DuplicatingThread *sourceThread, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount); |
| virtual ~OutputTrack(); |
| |
| virtual status_t start(pid_t tid); |
| virtual void stop(); |
| bool write(int16_t* data, uint32_t frames); |
| bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } |
| bool isActive() const { return mActive; } |
| const wp<ThreadBase>& thread() const { return mThread; } |
| |
| private: |
| |
| enum { |
| NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value |
| }; |
| |
| status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); |
| void clearBufferQueue(); |
| |
| // Maximum number of pending buffers allocated by OutputTrack::write() |
| static const uint8_t kMaxOverFlowBuffers = 10; |
| |
| Vector < Buffer* > mBufferQueue; |
| AudioBufferProvider::Buffer mOutBuffer; |
| bool mActive; |
| DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() |
| }; // end of OutputTrack |
| |
| PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, |
| audio_io_handle_t id, uint32_t device, type_t type); |
| virtual ~PlaybackThread(); |
| |
| status_t dump(int fd, const Vector<String16>& args); |
| |
| // Thread virtuals |
| virtual status_t readyToRun(); |
| virtual bool threadLoop(); |
| |
| // RefBase |
| virtual void onFirstRef(); |
| |
| protected: |
| // Code snippets that were lifted up out of threadLoop() |
| virtual void threadLoop_mix() = 0; |
| virtual void threadLoop_sleepTime() = 0; |
| virtual void threadLoop_write(); |
| virtual void threadLoop_standby(); |
| |
| // prepareTracks_l reads and writes mActiveTracks, and also returns the |
| // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is |
| // responsible for clearing or destroying this Vector later on, when it |
| // is safe to do so. That will drop the final ref count and destroy the tracks. |
| virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; |
| |
| public: |
| |
| virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } |
| |
| // return estimated latency in milliseconds, as reported by HAL |
| uint32_t latency() const; |
| |
| void setMasterVolume(float value); |
| void setMasterMute(bool muted); |
| |
| void setStreamVolume(audio_stream_type_t stream, float value); |
| void setStreamMute(audio_stream_type_t stream, bool muted); |
| |
| float streamVolume(audio_stream_type_t stream) const; |
| |
| sp<Track> createTrack_l( |
| const sp<AudioFlinger::Client>& client, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId, |
| bool isTimed, |
| status_t *status); |
| |
| AudioStreamOut* getOutput() const; |
| AudioStreamOut* clearOutput(); |
| virtual audio_stream_t* stream(); |
| |
| void suspend() { mSuspended++; } |
| void restore() { if (mSuspended > 0) mSuspended--; } |
| bool isSuspended() const { return (mSuspended > 0); } |
| virtual String8 getParameters(const String8& keys); |
| virtual void audioConfigChanged_l(int event, int param = 0); |
| status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); |
| int16_t *mixBuffer() const { return mMixBuffer; }; |
| |
| virtual void detachAuxEffect_l(int effectId); |
| status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, |
| int EffectId); |
| status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, |
| int EffectId); |
| |
| virtual status_t addEffectChain_l(const sp<EffectChain>& chain); |
| virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); |
| virtual uint32_t hasAudioSession(int sessionId); |
| virtual uint32_t getStrategyForSession_l(int sessionId); |
| |
| void setStreamValid(audio_stream_type_t streamType, bool valid); |
| |
| protected: |
| int16_t* mMixBuffer; |
| uint32_t mSuspended; // suspend count, > 0 means suspended |
| int mBytesWritten; |
| private: |
| // mMasterMute is in both PlaybackThread and in AudioFlinger. When a |
| // PlaybackThread needs to find out if master-muted, it checks it's local |
| // copy rather than the one in AudioFlinger. This optimization saves a lock. |
| bool mMasterMute; |
| void setMasterMute_l(bool muted) { mMasterMute = muted; } |
| protected: |
| SortedVector< wp<Track> > mActiveTracks; |
| |
| 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; |
| |
| // Code snippets that are temporarily lifted up out of threadLoop() until the merge |
| void checkSilentMode_l(); |
| |
| // Non-trivial for DUPLICATING only |
| virtual void saveOutputTracks() { } |
| virtual void clearOutputTracks() { } |
| |
| // Cache various calculated values, at threadLoop() entry and after a parameter change |
| virtual void cacheParameters_l(); |
| |
| private: |
| |
| friend class AudioFlinger; // for numerous |
| |
| PlaybackThread(const Client&); |
| PlaybackThread& operator = (const PlaybackThread&); |
| |
| status_t addTrack_l(const sp<Track>& track); |
| void destroyTrack_l(const sp<Track>& track); |
| void removeTrack_l(const sp<Track>& track); |
| |
| void readOutputParameters(); |
| |
| virtual status_t dumpInternals(int fd, const Vector<String16>& args); |
| status_t dumpTracks(int fd, const Vector<String16>& args); |
| |
| SortedVector< sp<Track> > mTracks; |
| // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread |
| stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; |
| AudioStreamOut *mOutput; |
| float mMasterVolume; |
| nsecs_t mLastWriteTime; |
| int mNumWrites; |
| int mNumDelayedWrites; |
| bool mInWrite; |
| |
| // FIXME rename these former local variables of threadLoop to standard "m" names |
| nsecs_t standbyTime; |
| size_t mixBufferSize; |
| |
| // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() |
| uint32_t activeSleepTime; |
| uint32_t idleSleepTime; |
| |
| uint32_t sleepTime; |
| |
| // mixer status returned by prepareTracks_l() |
| mixer_state mMixerStatus; // current cycle |
| mixer_state mPrevMixerStatus; // previous cycle |
| |
| // FIXME move these declarations into the specific sub-class that needs them |
| // MIXER only |
| bool longStandbyExit; |
| uint32_t sleepTimeShift; |
| |
| // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value |
| nsecs_t standbyDelay; |
| |
| // MIXER only |
| nsecs_t maxPeriod; |
| |
| // DUPLICATING only |
| uint32_t writeFrames; |
| }; |
| |
| class MixerThread : public PlaybackThread { |
| public: |
| MixerThread (const sp<AudioFlinger>& audioFlinger, |
| AudioStreamOut* output, |
| audio_io_handle_t id, |
| uint32_t device, |
| type_t type = MIXER); |
| virtual ~MixerThread(); |
| |
| // Thread virtuals |
| |
| void invalidateTracks(audio_stream_type_t streamType); |
| virtual bool checkForNewParameters_l(); |
| virtual status_t dumpInternals(int fd, const Vector<String16>& args); |
| |
| protected: |
| 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 void cacheParameters_l(); |
| |
| // threadLoop snippets |
| virtual void threadLoop_mix(); |
| virtual void threadLoop_sleepTime(); |
| |
| AudioMixer* mAudioMixer; |
| }; |
| |
| class DirectOutputThread : public PlaybackThread { |
| public: |
| |
| DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, |
| audio_io_handle_t id, uint32_t device); |
| virtual ~DirectOutputThread(); |
| |
| // Thread virtuals |
| |
| virtual bool checkForNewParameters_l(); |
| |
| protected: |
| virtual int getTrackName_l(); |
| virtual void deleteTrackName_l(int name); |
| virtual uint32_t activeSleepTimeUs(); |
| virtual uint32_t idleSleepTimeUs(); |
| virtual uint32_t suspendSleepTimeUs(); |
| virtual void cacheParameters_l(); |
| |
| // threadLoop snippets |
| virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); |
| virtual void threadLoop_mix(); |
| virtual void threadLoop_sleepTime(); |
| |
| // volumes last sent to audio HAL with stream->set_volume() |
| // FIXME use standard representation and names |
| float mLeftVolFloat; |
| float mRightVolFloat; |
| uint16_t mLeftVolShort; |
| uint16_t mRightVolShort; |
| |
| // FIXME rename these former local variables of threadLoop to standard names |
| // next 3 were local to the while !exitingPending loop |
| bool rampVolume; |
| uint16_t leftVol; |
| uint16_t rightVol; |
| |
| private: |
| // prepareTracks_l() tells threadLoop_mix() the name of the single active track |
| sp<Track> mActiveTrack; |
| }; |
| |
| class DuplicatingThread : public MixerThread { |
| public: |
| DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, |
| audio_io_handle_t id); |
| virtual ~DuplicatingThread(); |
| |
| // Thread virtuals |
| void addOutputTrack(MixerThread* thread); |
| void removeOutputTrack(MixerThread* thread); |
| uint32_t waitTimeMs() { return mWaitTimeMs; } |
| protected: |
| virtual uint32_t activeSleepTimeUs(); |
| |
| private: |
| bool outputsReady(const SortedVector<sp<OutputTrack> > &outputTracks); |
| protected: |
| // threadLoop snippets |
| virtual void threadLoop_mix(); |
| virtual void threadLoop_sleepTime(); |
| virtual void threadLoop_write(); |
| virtual void threadLoop_standby(); |
| virtual void cacheParameters_l(); |
| |
| private: |
| // called from threadLoop, addOutputTrack, removeOutputTrack |
| virtual void updateWaitTime_l(); |
| protected: |
| virtual void saveOutputTracks(); |
| virtual void clearOutputTracks(); |
| private: |
| |
| uint32_t mWaitTimeMs; |
| SortedVector < sp<OutputTrack> > outputTracks; |
| SortedVector < sp<OutputTrack> > mOutputTracks; |
| }; |
| |
| PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; |
| MixerThread *checkMixerThread_l(audio_io_handle_t output) const; |
| RecordThread *checkRecordThread_l(audio_io_handle_t input) const; |
| // no range check, AudioFlinger::mLock held |
| bool streamMute_l(audio_stream_type_t stream) const |
| { return mStreamTypes[stream].mute; } |
| // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held |
| float streamVolume_l(audio_stream_type_t stream) const |
| { return mStreamTypes[stream].volume; } |
| void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); |
| |
| // allocate an audio_io_handle_t, session ID, or effect ID |
| uint32_t nextUniqueId(); |
| |
| status_t moveEffectChain_l(int sessionId, |
| PlaybackThread *srcThread, |
| PlaybackThread *dstThread, |
| bool reRegister); |
| // return thread associated with primary hardware device, or NULL |
| PlaybackThread *primaryPlaybackThread_l() const; |
| uint32_t primaryOutputDevice_l() const; |
| |
| // server side of the client's IAudioTrack |
| class TrackHandle : public android::BnAudioTrack { |
| public: |
| TrackHandle(const sp<PlaybackThread::Track>& track); |
| virtual ~TrackHandle(); |
| virtual sp<IMemory> getCblk() const; |
| virtual status_t start(pid_t tid); |
| virtual void stop(); |
| virtual void flush(); |
| virtual void mute(bool); |
| virtual void pause(); |
| virtual status_t attachAuxEffect(int effectId); |
| virtual status_t allocateTimedBuffer(size_t size, |
| sp<IMemory>* buffer); |
| virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, |
| int64_t pts); |
| virtual status_t setMediaTimeTransform(const LinearTransform& xform, |
| int target); |
| virtual status_t onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); |
| private: |
| const sp<PlaybackThread::Track> mTrack; |
| }; |
| |
| void removeClient_l(pid_t pid); |
| void removeNotificationClient(pid_t pid); |
| |
| |
| // record thread |
| class RecordThread : public ThreadBase, public AudioBufferProvider |
| { |
| public: |
| |
| // record track |
| class RecordTrack : public TrackBase { |
| public: |
| RecordTrack(RecordThread *thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| audio_format_t format, |
| uint32_t channelMask, |
| int frameCount, |
| int sessionId); |
| virtual ~RecordTrack(); |
| |
| virtual status_t start(pid_t tid); |
| virtual void stop(); |
| |
| bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } |
| bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } |
| |
| void dump(char* buffer, size_t size); |
| |
| private: |
| friend class AudioFlinger; // for mState |
| |
| RecordTrack(const RecordTrack&); |
| RecordTrack& operator = (const RecordTrack&); |
| |
| // AudioBufferProvider interface |
| virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); |
| // releaseBuffer() not overridden |
| |
| bool mOverflow; |
| }; |
| |
| |
| RecordThread(const sp<AudioFlinger>& audioFlinger, |
| AudioStreamIn *input, |
| uint32_t sampleRate, |
| uint32_t channels, |
| audio_io_handle_t id, |
| uint32_t device); |
| virtual ~RecordThread(); |
| |
| // Thread |
| 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, |
| int channelMask, |
| int frameCount, |
| int sessionId, |
| status_t *status); |
| |
| status_t start(RecordTrack* recordTrack); |
| status_t start(RecordTrack* recordTrack, pid_t tid); |
| void stop(RecordTrack* recordTrack); |
| status_t dump(int fd, const Vector<String16>& args); |
| AudioStreamIn* getInput() const; |
| AudioStreamIn* clearInput(); |
| virtual audio_stream_t* stream(); |
| |
| // 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(); |
| |
| virtual status_t addEffectChain_l(const sp<EffectChain>& chain); |
| virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); |
| virtual uint32_t hasAudioSession(int sessionId); |
| RecordTrack* track(); |
| |
| private: |
| RecordThread(); |
| AudioStreamIn *mInput; |
| RecordTrack* mTrack; |
| sp<RecordTrack> mActiveTrack; |
| Condition mStartStopCond; |
| AudioResampler *mResampler; |
| int32_t *mRsmpOutBuffer; |
| int16_t *mRsmpInBuffer; |
| size_t mRsmpInIndex; |
| size_t mInputBytes; |
| const int mReqChannelCount; |
| const uint32_t mReqSampleRate; |
| ssize_t mBytesRead; |
| }; |
| |
| // server side of the client's IAudioRecord |
| class RecordHandle : public android::BnAudioRecord { |
| public: |
| RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); |
| virtual ~RecordHandle(); |
| virtual sp<IMemory> getCblk() const; |
| virtual status_t start(pid_t tid); |
| virtual void stop(); |
| virtual status_t onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); |
| private: |
| const sp<RecordThread::RecordTrack> mRecordTrack; |
| }; |
| |
| //--- Audio Effect Management |
| |
| // EffectModule and EffectChain classes both have their own mutex to protect |
| // state changes or resource modifications. Always respect the following order |
| // if multiple mutexes must be acquired to avoid cross deadlock: |
| // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule |
| |
| // The EffectModule class is a wrapper object controlling the effect engine implementation |
| // in the effect library. It prevents concurrent calls to process() and command() functions |
| // from different client threads. It keeps a list of EffectHandle objects corresponding |
| // to all client applications using this effect and notifies applications of effect state, |
| // control or parameter changes. It manages the activation state machine to send appropriate |
| // reset, enable, disable commands to effect engine and provide volume |
| // ramping when effects are activated/deactivated. |
| // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by |
| // the attached track(s) to accumulate their auxiliary channel. |
| class EffectModule: public RefBase { |
| public: |
| EffectModule(ThreadBase *thread, |
| const wp<AudioFlinger::EffectChain>& chain, |
| effect_descriptor_t *desc, |
| int id, |
| int sessionId); |
| virtual ~EffectModule(); |
| |
| enum effect_state { |
| IDLE, |
| RESTART, |
| STARTING, |
| ACTIVE, |
| STOPPING, |
| STOPPED, |
| DESTROYED |
| }; |
| |
| int id() const { return mId; } |
| void process(); |
| void updateState(); |
| status_t command(uint32_t cmdCode, |
| uint32_t cmdSize, |
| void *pCmdData, |
| uint32_t *replySize, |
| void *pReplyData); |
| |
| void reset_l(); |
| status_t configure(); |
| status_t init(); |
| effect_state state() const { |
| return mState; |
| } |
| uint32_t status() { |
| return mStatus; |
| } |
| int sessionId() const { |
| return mSessionId; |
| } |
| status_t setEnabled(bool enabled); |
| bool isEnabled() const; |
| bool isProcessEnabled() const; |
| |
| void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } |
| int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } |
| void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } |
| int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } |
| void setChain(const wp<EffectChain>& chain) { mChain = chain; } |
| void setThread(const wp<ThreadBase>& thread) { mThread = thread; } |
| const wp<ThreadBase>& thread() { return mThread; } |
| |
| status_t addHandle(const sp<EffectHandle>& handle); |
| void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); |
| size_t removeHandle (const wp<EffectHandle>& handle); |
| |
| effect_descriptor_t& desc() { return mDescriptor; } |
| wp<EffectChain>& chain() { return mChain; } |
| |
| status_t setDevice(uint32_t device); |
| status_t setVolume(uint32_t *left, uint32_t *right, bool controller); |
| status_t setMode(audio_mode_t mode); |
| status_t start(); |
| status_t stop(); |
| void setSuspended(bool suspended); |
| bool suspended() const; |
| |
| sp<EffectHandle> controlHandle(); |
| |
| bool isPinned() const { return mPinned; } |
| void unPin() { mPinned = false; } |
| |
| status_t dump(int fd, const Vector<String16>& args); |
| |
| protected: |
| friend class AudioFlinger; // for mHandles |
| bool mPinned; |
| |
| // Maximum time allocated to effect engines to complete the turn off sequence |
| static const uint32_t MAX_DISABLE_TIME_MS = 10000; |
| |
| EffectModule(const EffectModule&); |
| EffectModule& operator = (const EffectModule&); |
| |
| status_t start_l(); |
| status_t stop_l(); |
| |
| mutable Mutex mLock; // mutex for process, commands and handles list protection |
| wp<ThreadBase> mThread; // parent thread |
| wp<EffectChain> mChain; // parent effect chain |
| int mId; // this instance unique ID |
| int mSessionId; // audio session ID |
| effect_descriptor_t mDescriptor;// effect descriptor received from effect engine |
| effect_config_t mConfig; // input and output audio configuration |
| effect_handle_t mEffectInterface; // Effect module C API |
| status_t mStatus; // initialization status |
| effect_state mState; // current activation state |
| Vector< wp<EffectHandle> > mHandles; // list of client handles |
| // First handle in mHandles has highest priority and controls the effect module |
| uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after |
| // sending disable command. |
| uint32_t mDisableWaitCnt; // current process() calls count during disable period. |
| bool mSuspended; // effect is suspended: temporarily disabled by framework |
| }; |
| |
| // The EffectHandle class implements the IEffect interface. It provides resources |
| // to receive parameter updates, keeps track of effect control |
| // ownership and state and has a pointer to the EffectModule object it is controlling. |
| // There is one EffectHandle object for each application controlling (or using) |
| // an effect module. |
| // The EffectHandle is obtained by calling AudioFlinger::createEffect(). |
| class EffectHandle: public android::BnEffect { |
| public: |
| |
| EffectHandle(const sp<EffectModule>& effect, |
| const sp<AudioFlinger::Client>& client, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority); |
| virtual ~EffectHandle(); |
| |
| // IEffect |
| virtual status_t enable(); |
| virtual status_t disable(); |
| virtual status_t command(uint32_t cmdCode, |
| uint32_t cmdSize, |
| void *pCmdData, |
| uint32_t *replySize, |
| void *pReplyData); |
| virtual void disconnect(); |
| private: |
| void disconnect(bool unpinIfLast); |
| public: |
| virtual sp<IMemory> getCblk() const { return mCblkMemory; } |
| virtual status_t onTransact(uint32_t code, const Parcel& data, |
| Parcel* reply, uint32_t flags); |
| |
| |
| // Give or take control of effect module |
| // - hasControl: true if control is given, false if removed |
| // - signal: true client app should be signaled of change, false otherwise |
| // - enabled: state of the effect when control is passed |
| void setControl(bool hasControl, bool signal, bool enabled); |
| void commandExecuted(uint32_t cmdCode, |
| uint32_t cmdSize, |
| void *pCmdData, |
| uint32_t replySize, |
| void *pReplyData); |
| void setEnabled(bool enabled); |
| bool enabled() const { return mEnabled; } |
| |
| // Getters |
| int id() const { return mEffect->id(); } |
| int priority() const { return mPriority; } |
| bool hasControl() const { return mHasControl; } |
| sp<EffectModule> effect() const { return mEffect; } |
| |
| void dump(char* buffer, size_t size); |
| |
| protected: |
| friend class AudioFlinger; // for mEffect, mHasControl, mEnabled |
| EffectHandle(const EffectHandle&); |
| EffectHandle& operator =(const EffectHandle&); |
| |
| sp<EffectModule> mEffect; // pointer to controlled EffectModule |
| sp<IEffectClient> mEffectClient; // callback interface for client notifications |
| /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() |
| sp<IMemory> mCblkMemory; // shared memory for control block |
| effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory |
| uint8_t* mBuffer; // pointer to parameter area in shared memory |
| int mPriority; // client application priority to control the effect |
| bool mHasControl; // true if this handle is controlling the effect |
| bool mEnabled; // cached enable state: needed when the effect is |
| // restored after being suspended |
| }; |
| |
| // the EffectChain class represents a group of effects associated to one audio session. |
| // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). |
| // The EffecChain with session ID 0 contains global effects applied to the output mix. |
| // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) |
| // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding |
| // in the effect process order. When attached to a track (session ID != 0), it also provide it's own |
| // input buffer used by the track as accumulation buffer. |
| class EffectChain: public RefBase { |
| public: |
| EffectChain(const wp<ThreadBase>& wThread, int sessionId); |
| EffectChain(ThreadBase *thread, int sessionId); |
| virtual ~EffectChain(); |
| |
| // special key used for an entry in mSuspendedEffects keyed vector |
| // corresponding to a suspend all request. |
| static const int kKeyForSuspendAll = 0; |
| |
| // minimum duration during which we force calling effect process when last track on |
| // a session is stopped or removed to allow effect tail to be rendered |
| static const int kProcessTailDurationMs = 1000; |
| |
| void process_l(); |
| |
| void lock() { |
| mLock.lock(); |
| } |
| void unlock() { |
| mLock.unlock(); |
| } |
| |
| status_t addEffect_l(const sp<EffectModule>& handle); |
| size_t removeEffect_l(const sp<EffectModule>& handle); |
| |
| int sessionId() const { return mSessionId; } |
| void setSessionId(int sessionId) { mSessionId = sessionId; } |
| |
| sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); |
| sp<EffectModule> getEffectFromId_l(int id); |
| sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); |
| bool setVolume_l(uint32_t *left, uint32_t *right); |
| void setDevice_l(uint32_t device); |
| void setMode_l(audio_mode_t mode); |
| |
| void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { |
| mInBuffer = buffer; |
| mOwnInBuffer = ownsBuffer; |
| } |
| int16_t *inBuffer() const { |
| return mInBuffer; |
| } |
| void setOutBuffer(int16_t *buffer) { |
| mOutBuffer = buffer; |
| } |
| int16_t *outBuffer() const { |
| return mOutBuffer; |
| } |
| |
| void incTrackCnt() { android_atomic_inc(&mTrackCnt); } |
| void decTrackCnt() { android_atomic_dec(&mTrackCnt); } |
| int32_t trackCnt() const { return mTrackCnt;} |
| |
| void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); |
| mTailBufferCount = mMaxTailBuffers; } |
| void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } |
| int32_t activeTrackCnt() const { return mActiveTrackCnt;} |
| |
| uint32_t strategy() const { return mStrategy; } |
| void setStrategy(uint32_t strategy) |
| { mStrategy = strategy; } |
| |
| // suspend effect of the given type |
| void setEffectSuspended_l(const effect_uuid_t *type, |
| bool suspend); |
| // suspend all eligible effects |
| void setEffectSuspendedAll_l(bool suspend); |
| // check if effects should be suspend or restored when a given effect is enable or disabled |
| void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, |
| bool enabled); |
| |
| status_t dump(int fd, const Vector<String16>& args); |
| |
| protected: |
| friend class AudioFlinger; // for mThread, mEffects |
| EffectChain(const EffectChain&); |
| EffectChain& operator =(const EffectChain&); |
| |
| class SuspendedEffectDesc : public RefBase { |
| public: |
| SuspendedEffectDesc() : mRefCount(0) {} |
| |
| int mRefCount; |
| effect_uuid_t mType; |
| wp<EffectModule> mEffect; |
| }; |
| |
| // get a list of effect modules to suspend when an effect of the type |
| // passed is enabled. |
| void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); |
| |
| // get an effect module if it is currently enable |
| sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); |
| // true if the effect whose descriptor is passed can be suspended |
| // OEMs can modify the rules implemented in this method to exclude specific effect |
| // types or implementations from the suspend/restore mechanism. |
| bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); |
| |
| wp<ThreadBase> mThread; // parent mixer thread |
| Mutex mLock; // mutex protecting effect list |
| Vector<sp<EffectModule> > mEffects; // list of effect modules |
| int mSessionId; // audio session ID |
| int16_t *mInBuffer; // chain input buffer |
| int16_t *mOutBuffer; // chain output buffer |
| volatile int32_t mActiveTrackCnt; // number of active tracks connected |
| volatile int32_t mTrackCnt; // number of tracks connected |
| int32_t mTailBufferCount; // current effect tail buffer count |
| int32_t mMaxTailBuffers; // maximum effect tail buffers |
| bool mOwnInBuffer; // true if the chain owns its input buffer |
| int mVolumeCtrlIdx; // index of insert effect having control over volume |
| uint32_t mLeftVolume; // previous volume on left channel |
| uint32_t mRightVolume; // previous volume on right channel |
| uint32_t mNewLeftVolume; // new volume on left channel |
| uint32_t mNewRightVolume; // new volume on right channel |
| uint32_t mStrategy; // strategy for this effect chain |
| // mSuspendedEffects lists all effect currently suspended in the chain |
| // use effect type UUID timelow field as key. There is no real risk of identical |
| // timeLow fields among effect type UUIDs. |
| KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; |
| }; |
| |
| // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. |
| // For emphasis, we could also make all pointers to them be "const *", |
| // but that would clutter the code unnecessarily. |
| |
| struct AudioStreamOut { |
| audio_hw_device_t* const hwDev; |
| audio_stream_out_t* const stream; |
| |
| AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : |
| hwDev(dev), stream(out) {} |
| }; |
| |
| struct AudioStreamIn { |
| audio_hw_device_t* const hwDev; |
| audio_stream_in_t* const stream; |
| |
| AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : |
| hwDev(dev), stream(in) {} |
| }; |
| |
| // for mAudioSessionRefs only |
| struct AudioSessionRef { |
| AudioSessionRef(int sessionid, pid_t pid) : |
| mSessionid(sessionid), mPid(pid), mCnt(1) {} |
| const int mSessionid; |
| const pid_t mPid; |
| int mCnt; |
| }; |
| |
| enum master_volume_support { |
| // MVS_NONE: |
| // Audio HAL has no support for master volume, either setting or |
| // getting. All master volume control must be implemented in SW by the |
| // AudioFlinger mixing core. |
| MVS_NONE, |
| |
| // MVS_SETONLY: |
| // Audio HAL has support for setting master volume, but not for getting |
| // master volume (original HAL design did not include a getter). |
| // AudioFlinger needs to keep track of the last set master volume in |
| // addition to needing to set an initial, default, master volume at HAL |
| // load time. |
| MVS_SETONLY, |
| |
| // MVS_FULL: |
| // Audio HAL has support both for setting and getting master volume. |
| // AudioFlinger should send all set and get master volume requests |
| // directly to the HAL. |
| MVS_FULL, |
| }; |
| |
| mutable Mutex mLock; |
| |
| DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() |
| |
| mutable Mutex mHardwareLock; |
| |
| // These two fields are immutable after onFirstRef(), so no lock needed to access |
| audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL |
| Vector<audio_hw_device_t*> mAudioHwDevs; |
| |
| // for dump, indicates which hardware operation is currently in progress (but not stream ops) |
| enum hardware_call_state { |
| AUDIO_HW_IDLE = 0, // no operation in progress |
| AUDIO_HW_INIT, // init_check |
| AUDIO_HW_OUTPUT_OPEN, // open_output_stream |
| AUDIO_HW_OUTPUT_CLOSE, // unused |
| AUDIO_HW_INPUT_OPEN, // unused |
| AUDIO_HW_INPUT_CLOSE, // unused |
| AUDIO_HW_STANDBY, // unused |
| AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume |
| AUDIO_HW_GET_ROUTING, // unused |
| AUDIO_HW_SET_ROUTING, // unused |
| AUDIO_HW_GET_MODE, // unused |
| AUDIO_HW_SET_MODE, // set_mode |
| AUDIO_HW_GET_MIC_MUTE, // get_mic_mute |
| AUDIO_HW_SET_MIC_MUTE, // set_mic_mute |
| AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume |
| AUDIO_HW_SET_PARAMETER, // set_parameters |
| AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size |
| AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume |
| AUDIO_HW_GET_PARAMETER, // get_parameters |
| }; |
| |
| mutable hardware_call_state mHardwareStatus; // for dump only |
| |
| |
| DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; |
| stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; |
| |
| // both are protected by mLock |
| float mMasterVolume; |
| float mMasterVolumeSW; |
| master_volume_support mMasterVolumeSupportLvl; |
| bool mMasterMute; |
| |
| DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; |
| |
| DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; |
| volatile int32_t mNextUniqueId; // updated by android_atomic_inc |
| audio_mode_t mMode; |
| bool mBtNrecIsOff; |
| |
| // protected by mLock |
| Vector<AudioSessionRef*> mAudioSessionRefs; |
| |
| float masterVolume_l() const; |
| float masterVolumeSW_l() const { return mMasterVolumeSW; } |
| bool masterMute_l() const { return mMasterMute; } |
| |
| private: |
| sp<Client> registerPid_l(pid_t pid); // always returns non-0 |
| |
| }; |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIO_FLINGER_H |