| /* |
| ** |
| ** 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 "Configuration.h" |
| #include <stdint.h> |
| #include <sys/types.h> |
| #include <limits.h> |
| |
| #include <common_time/cc_helper.h> |
| |
| #include <cutils/compiler.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 <hardware/audio_policy.h> |
| |
| #include <media/AudioBufferProvider.h> |
| #include <media/ExtendedAudioBufferProvider.h> |
| |
| #include "FastCapture.h" |
| #include "FastMixer.h" |
| #include <media/nbaio/NBAIO.h> |
| #include "AudioWatchdog.h" |
| #include "AudioMixer.h" |
| #include "AudioStreamOut.h" |
| #include "SpdifStreamOut.h" |
| #include "AudioHwDevice.h" |
| |
| #include <powermanager/IPowerManager.h> |
| |
| #include <media/nbaio/NBLog.h> |
| #include <private/media/AudioTrackShared.h> |
| |
| namespace android { |
| |
| struct audio_track_cblk_t; |
| struct effect_param_cblk_t; |
| class AudioMixer; |
| class AudioBuffer; |
| class AudioResampler; |
| class FastMixer; |
| class PassthruBufferProvider; |
| class ServerProxy; |
| |
| // ---------------------------------------------------------------------------- |
| |
| // The macro FCC_2 highlights some (but not all) places where there are are 2-channel assumptions. |
| // This is typically due to legacy implementation of stereo input or output. |
| // Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. |
| #define FCC_2 2 // FCC_2 = Fixed Channel Count 2 |
| // The macro FCC_8 highlights places where there are 8-channel assumptions. |
| // This is typically due to audio mixer and resampler limitations. |
| #define FCC_8 8 // FCC_8 = Fixed Channel Count 8 |
| |
| static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); |
| |
| #define INCLUDING_FROM_AUDIOFLINGER_H |
| |
| class AudioFlinger : |
| public BinderService<AudioFlinger>, |
| public BnAudioFlinger |
| { |
| friend class BinderService<AudioFlinger>; // for AudioFlinger() |
| public: |
| static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } |
| |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| |
| // IAudioFlinger interface, in binder opcode order |
| virtual sp<IAudioTrack> createTrack( |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t *pFrameCount, |
| IAudioFlinger::track_flags_t *flags, |
| const sp<IMemory>& sharedBuffer, |
| audio_io_handle_t output, |
| pid_t tid, |
| int *sessionId, |
| int clientUid, |
| status_t *status /*non-NULL*/); |
| |
| virtual sp<IAudioRecord> openRecord( |
| audio_io_handle_t input, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| const String16& opPackageName, |
| size_t *pFrameCount, |
| IAudioFlinger::track_flags_t *flags, |
| pid_t tid, |
| int *sessionId, |
| size_t *notificationFrames, |
| sp<IMemory>& cblk, |
| sp<IMemory>& buffers, |
| status_t *status /*non-NULL*/); |
| |
| virtual uint32_t sampleRate(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 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, |
| audio_channel_mask_t channelMask) const; |
| |
| virtual status_t openOutput(audio_module_handle_t module, |
| audio_io_handle_t *output, |
| audio_config_t *config, |
| audio_devices_t *devices, |
| const String8& address, |
| uint32_t *latencyMs, |
| audio_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 status_t openInput(audio_module_handle_t module, |
| audio_io_handle_t *input, |
| audio_config_t *config, |
| audio_devices_t *device, |
| const String8& address, |
| audio_source_t source, |
| audio_input_flags_t flags); |
| |
| virtual status_t closeInput(audio_io_handle_t input); |
| |
| virtual status_t invalidateStream(audio_stream_type_t stream); |
| |
| virtual status_t setVoiceVolume(float volume); |
| |
| virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, |
| audio_io_handle_t output) const; |
| |
| virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; |
| |
| virtual audio_unique_id_t newAudioUniqueId(); |
| |
| virtual void acquireAudioSessionId(int audioSession, pid_t pid); |
| |
| virtual void releaseAudioSessionId(int audioSession, pid_t pid); |
| |
| 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( |
| effect_descriptor_t *pDesc, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority, |
| audio_io_handle_t io, |
| int sessionId, |
| const String16& opPackageName, |
| status_t *status /*non-NULL*/, |
| int *id, |
| int *enabled); |
| |
| virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, |
| audio_io_handle_t dstOutput); |
| |
| virtual audio_module_handle_t loadHwModule(const char *name); |
| |
| virtual uint32_t getPrimaryOutputSamplingRate(); |
| virtual size_t getPrimaryOutputFrameCount(); |
| |
| virtual status_t setLowRamDevice(bool isLowRamDevice); |
| |
| /* List available audio ports and their attributes */ |
| virtual status_t listAudioPorts(unsigned int *num_ports, |
| struct audio_port *ports); |
| |
| /* Get attributes for a given audio port */ |
| virtual status_t getAudioPort(struct audio_port *port); |
| |
| /* Create an audio patch between several source and sink ports */ |
| virtual status_t createAudioPatch(const struct audio_patch *patch, |
| audio_patch_handle_t *handle); |
| |
| /* Release an audio patch */ |
| virtual status_t releaseAudioPatch(audio_patch_handle_t handle); |
| |
| /* List existing audio patches */ |
| virtual status_t listAudioPatches(unsigned int *num_patches, |
| struct audio_patch *patches); |
| |
| /* Set audio port configuration */ |
| virtual status_t setAudioPortConfig(const struct audio_port_config *config); |
| |
| /* Get the HW synchronization source used for an audio session */ |
| virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); |
| |
| virtual status_t onTransact( |
| uint32_t code, |
| const Parcel& data, |
| Parcel* reply, |
| uint32_t flags); |
| |
| // end of IAudioFlinger interface |
| |
| sp<NBLog::Writer> newWriter_l(size_t size, const char *name); |
| void unregisterWriter(const sp<NBLog::Writer>& writer); |
| private: |
| static const size_t kLogMemorySize = 40 * 1024; |
| sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled |
| // When a log writer is unregistered, it is done lazily so that media.log can continue to see it |
| // for as long as possible. The memory is only freed when it is needed for another log writer. |
| Vector< sp<NBLog::Writer> > mUnregisteredWriters; |
| Mutex mUnregisteredWritersLock; |
| public: |
| |
| class SyncEvent; |
| |
| typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; |
| |
| class SyncEvent : public RefBase { |
| public: |
| SyncEvent(AudioSystem::sync_event_t type, |
| int triggerSession, |
| int listenerSession, |
| sync_event_callback_t callBack, |
| wp<RefBase> cookie) |
| : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), |
| mCallback(callBack), mCookie(cookie) |
| {} |
| |
| virtual ~SyncEvent() {} |
| |
| void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } |
| bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } |
| void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } |
| AudioSystem::sync_event_t type() const { return mType; } |
| int triggerSession() const { return mTriggerSession; } |
| int listenerSession() const { return mListenerSession; } |
| wp<RefBase> cookie() const { return mCookie; } |
| |
| private: |
| const AudioSystem::sync_event_t mType; |
| const int mTriggerSession; |
| const int mListenerSession; |
| sync_event_callback_t mCallback; |
| const wp<RefBase> mCookie; |
| mutable Mutex mLock; |
| }; |
| |
| sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, |
| int triggerSession, |
| int listenerSession, |
| sync_event_callback_t callBack, |
| wp<RefBase> cookie); |
| |
| private: |
| |
| audio_mode_t getMode() const { return mMode; } |
| |
| bool btNrecIsOff() const { return mBtNrecIsOff; } |
| |
| AudioFlinger() ANDROID_API; |
| 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(); |
| |
| AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, |
| audio_devices_t devices); |
| void purgeStaleEffects_l(); |
| |
| // Set kEnableExtendedChannels to true to enable greater than stereo output |
| // for the MixerThread and device sink. Number of channels allowed is |
| // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. |
| static const bool kEnableExtendedChannels = true; |
| |
| // Returns true if channel mask is permitted for the PCM sink in the MixerThread |
| static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { |
| switch (audio_channel_mask_get_representation(channelMask)) { |
| case AUDIO_CHANNEL_REPRESENTATION_POSITION: { |
| uint32_t channelCount = FCC_2; // stereo is default |
| if (kEnableExtendedChannels) { |
| channelCount = audio_channel_count_from_out_mask(channelMask); |
| if (channelCount < FCC_2 // mono is not supported at this time |
| || channelCount > AudioMixer::MAX_NUM_CHANNELS) { |
| return false; |
| } |
| } |
| // check that channelMask is the "canonical" one we expect for the channelCount. |
| return channelMask == audio_channel_out_mask_from_count(channelCount); |
| } |
| default: |
| return false; |
| } |
| } |
| |
| // Set kEnableExtendedPrecision to true to use extended precision in MixerThread |
| static const bool kEnableExtendedPrecision = true; |
| |
| // Returns true if format is permitted for the PCM sink in the MixerThread |
| static inline bool isValidPcmSinkFormat(audio_format_t format) { |
| switch (format) { |
| case AUDIO_FORMAT_PCM_16_BIT: |
| return true; |
| case AUDIO_FORMAT_PCM_FLOAT: |
| case AUDIO_FORMAT_PCM_24_BIT_PACKED: |
| case AUDIO_FORMAT_PCM_32_BIT: |
| case AUDIO_FORMAT_PCM_8_24_BIT: |
| return kEnableExtendedPrecision; |
| default: |
| return false; |
| } |
| } |
| |
| // 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; |
| |
| // incremented by 2 when screen state changes, bit 0 == 1 means "off" |
| // AudioFlinger::setParameters() updates, other threads read w/o lock |
| static uint32_t mScreenState; |
| |
| // Internal dump utilities. |
| static const int kDumpLockRetries = 50; |
| static const int kDumpLockSleepUs = 20000; |
| static bool dumpTryLock(Mutex& mutex); |
| void dumpPermissionDenial(int fd, const Vector<String16>& args); |
| void dumpClients(int fd, const Vector<String16>& args); |
| void 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 OffloadThread; |
| class DuplicatingThread; |
| class AsyncCallbackThread; |
| class Track; |
| class RecordTrack; |
| class EffectModule; |
| class EffectHandle; |
| class EffectChain; |
| |
| struct AudioStreamIn; |
| |
| struct stream_type_t { |
| stream_type_t() |
| : volume(1.0f), |
| mute(false) |
| { |
| } |
| float volume; |
| bool mute; |
| }; |
| |
| // --- PlaybackThread --- |
| |
| #include "Threads.h" |
| |
| #include "Effects.h" |
| |
| #include "PatchPanel.h" |
| |
| // 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(); |
| virtual void stop(); |
| virtual void flush(); |
| 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 setParameters(const String8& keyValuePairs); |
| virtual status_t getTimestamp(AudioTimestamp& timestamp); |
| virtual void signal(); // signal playback thread for a change in control block |
| |
| virtual status_t onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); |
| |
| private: |
| const sp<PlaybackThread::Track> mTrack; |
| }; |
| |
| // server side of the client's IAudioRecord |
| class RecordHandle : public android::BnAudioRecord { |
| public: |
| RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); |
| virtual ~RecordHandle(); |
| virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); |
| virtual void stop(); |
| virtual status_t onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); |
| private: |
| const sp<RecordThread::RecordTrack> mRecordTrack; |
| |
| // for use from destructor |
| void stop_nonvirtual(); |
| }; |
| |
| |
| 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; |
| sp<RecordThread> openInput_l(audio_module_handle_t module, |
| audio_io_handle_t *input, |
| audio_config_t *config, |
| audio_devices_t device, |
| const String8& address, |
| audio_source_t source, |
| audio_input_flags_t flags); |
| sp<PlaybackThread> openOutput_l(audio_module_handle_t module, |
| audio_io_handle_t *output, |
| audio_config_t *config, |
| audio_devices_t devices, |
| const String8& address, |
| audio_output_flags_t flags); |
| |
| void closeOutputFinish(sp<PlaybackThread> thread); |
| void closeInputFinish(sp<RecordThread> thread); |
| |
| // 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(int event, audio_io_handle_t ioHandle, const void *param2); |
| |
| // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. |
| // They all share the same ID space, but the namespaces are actually independent |
| // because there are separate KeyedVectors for each kind of ID. |
| // The return value is uint32_t, but is cast to signed for some IDs. |
| // FIXME This API does not handle rollover to zero (for unsigned IDs), |
| // or from positive to negative (for signed IDs). |
| // Thus it may fail by returning an ID of the wrong sign, |
| // or by returning a non-unique 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; |
| audio_devices_t primaryOutputDevice_l() const; |
| |
| sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); |
| |
| |
| void removeClient_l(pid_t pid); |
| void removeNotificationClient(pid_t pid); |
| bool isNonOffloadableGlobalEffectEnabled_l(); |
| void onNonOffloadableGlobalEffectEnable(); |
| |
| // Store an effect chain to mOrphanEffectChains keyed vector. |
| // Called when a thread exits and effects are still attached to it. |
| // If effects are later created on the same session, they will reuse the same |
| // effect chain and same instances in the effect library. |
| // return ALREADY_EXISTS if a chain with the same session already exists in |
| // mOrphanEffectChains. Note that this should never happen as there is only one |
| // chain for a given session and it is attached to only one thread at a time. |
| status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); |
| // Get an effect chain for the specified session in mOrphanEffectChains and remove |
| // it if found. Returns 0 if not found (this is the most common case). |
| sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); |
| // Called when the last effect handle on an effect instance is removed. If this |
| // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated |
| // and removed from mOrphanEffectChains if it does not contain any effect. |
| // Return true if the effect was found in mOrphanEffectChains, false otherwise. |
| bool updateOrphanEffectChains(const sp<EffectModule>& effect); |
| |
| |
| // AudioStreamIn is 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 AudioStreamIn { |
| AudioHwDevice* const audioHwDev; |
| audio_stream_in_t* const stream; |
| |
| audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } |
| |
| AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : |
| audioHwDev(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; |
| }; |
| |
| mutable Mutex mLock; |
| // protects mClients and mNotificationClients. |
| // must be locked after mLock and ThreadBase::mLock if both must be locked |
| // avoids acquiring AudioFlinger::mLock from inside thread loop. |
| mutable Mutex mClientLock; |
| // protected by mClientLock |
| DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() |
| |
| mutable Mutex mHardwareLock; |
| // NOTE: If both mLock and mHardwareLock mutexes must be held, |
| // always take mLock before mHardwareLock |
| |
| // These two fields are immutable after onFirstRef(), so no lock needed to access |
| AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL |
| DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> 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 |
| AUDIO_HW_SET_MASTER_MUTE, // set_master_mute |
| AUDIO_HW_GET_MASTER_MUTE, // get_master_mute |
| }; |
| |
| mutable hardware_call_state mHardwareStatus; // for dump only |
| |
| |
| DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; |
| stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; |
| |
| // member variables below are protected by mLock |
| float mMasterVolume; |
| bool mMasterMute; |
| // end of variables protected by mLock |
| |
| DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; |
| |
| // protected by mClientLock |
| DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; |
| |
| volatile int32_t mNextUniqueId; // updated by android_atomic_inc |
| // nextUniqueId() returns uint32_t, but this is declared int32_t |
| // because the atomic operations require an int32_t |
| |
| audio_mode_t mMode; |
| bool mBtNrecIsOff; |
| |
| // protected by mLock |
| Vector<AudioSessionRef*> mAudioSessionRefs; |
| |
| float masterVolume_l() const; |
| bool masterMute_l() const; |
| audio_module_handle_t loadHwModule_l(const char *name); |
| |
| Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session |
| // to be created |
| |
| // Effect chains without a valid thread |
| DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; |
| |
| // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL |
| DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; |
| private: |
| sp<Client> registerPid(pid_t pid); // always returns non-0 |
| |
| // for use from destructor |
| status_t closeOutput_nonvirtual(audio_io_handle_t output); |
| void closeOutputInternal_l(sp<PlaybackThread> thread); |
| status_t closeInput_nonvirtual(audio_io_handle_t input); |
| void closeInputInternal_l(sp<RecordThread> thread); |
| void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); |
| |
| status_t checkStreamType(audio_stream_type_t stream) const; |
| |
| #ifdef TEE_SINK |
| // all record threads serially share a common tee sink, which is re-created on format change |
| sp<NBAIO_Sink> mRecordTeeSink; |
| sp<NBAIO_Source> mRecordTeeSource; |
| #endif |
| |
| public: |
| |
| #ifdef TEE_SINK |
| // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file |
| static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); |
| |
| // whether tee sink is enabled by property |
| static bool mTeeSinkInputEnabled; |
| static bool mTeeSinkOutputEnabled; |
| static bool mTeeSinkTrackEnabled; |
| |
| // runtime configured size of each tee sink pipe, in frames |
| static size_t mTeeSinkInputFrames; |
| static size_t mTeeSinkOutputFrames; |
| static size_t mTeeSinkTrackFrames; |
| |
| // compile-time default size of tee sink pipes, in frames |
| // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes |
| static const size_t kTeeSinkInputFramesDefault = 0x200000; |
| static const size_t kTeeSinkOutputFramesDefault = 0x200000; |
| static const size_t kTeeSinkTrackFramesDefault = 0x200000; |
| #endif |
| |
| // This method reads from a variable without mLock, but the variable is updated under mLock. So |
| // we might read a stale value, or a value that's inconsistent with respect to other variables. |
| // In this case, it's safe because the return value isn't used for making an important decision. |
| // The reason we don't want to take mLock is because it could block the caller for a long time. |
| bool isLowRamDevice() const { return mIsLowRamDevice; } |
| |
| private: |
| bool mIsLowRamDevice; |
| bool mIsDeviceTypeKnown; |
| nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled |
| |
| sp<PatchPanel> mPatchPanel; |
| |
| uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none |
| // protected by mHardwareLock |
| }; |
| |
| #undef INCLUDING_FROM_AUDIOFLINGER_H |
| |
| const char *formatToString(audio_format_t format); |
| String8 inputFlagsToString(audio_input_flags_t flags); |
| String8 outputFlagsToString(audio_output_flags_t flags); |
| String8 devicesToString(audio_devices_t devices); |
| const char *sourceToString(audio_source_t source); |
| |
| // ---------------------------------------------------------------------------- |
| |
| } // namespace android |
| |
| #endif // ANDROID_AUDIO_FLINGER_H |