/* * Copyright (C) 2008 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_AUDIORECORD_H #define ANDROID_AUDIORECORD_H #include #include #include #include #include #include #include #include #include #include #include #include "android/media/IAudioRecord.h" #include namespace android { // ---------------------------------------------------------------------------- struct audio_track_cblk_t; class AudioRecordClientProxy; // ---------------------------------------------------------------------------- class AudioRecord : public AudioSystem::AudioDeviceCallback { public: class Buffer { friend AudioRecord; public: size_t size() const { return mSize; } size_t getFrameCount() const { return frameCount; } uint8_t* data() const { return ui8; } // Leaving public for now to assist refactoring. This class will // be replaced. size_t frameCount; // number of sample frames corresponding to size; // on input to obtainBuffer() it is the number of frames desired // on output from obtainBuffer() it is the number of available // frames to be read // on input to releaseBuffer() it is currently ignored private: size_t mSize; // input/output in bytes == frameCount * frameSize // on input to obtainBuffer() it is ignored // on output from obtainBuffer() it is the number of available // bytes to be read, which is frameCount * frameSize // on input to releaseBuffer() it is the number of bytes to // release // FIXME This is redundant with respect to frameCount. Consider // removing size and making frameCount the primary field. union { void* raw; int16_t* i16; // signed 16-bit uint8_t* ui8; // unsigned 8-bit, offset by 0x80 // input to obtainBuffer(): unused, output: pointer to buffer }; uint32_t sequence; // IAudioRecord instance sequence number, as of obtainBuffer(). // It is set by obtainBuffer() and confirmed by releaseBuffer(). // Not "user-serviceable". // TODO Consider sp instead, or in addition to this. }; /* As a convenience, if a callback is supplied, a handler thread * is automatically created with the appropriate priority. This thread * invokes the callback when a new buffer becomes available or various conditions occur. * Parameters: * * event: type of event notified (see enum AudioRecord::event_type). * user: Pointer to context for use by the callback receiver. * info: Pointer to optional parameter according to event type: * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read * more bytes than indicated by 'size' field and update 'size' if * fewer bytes are consumed. * - EVENT_OVERRUN: unused. * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. * - EVENT_NEW_IAUDIORECORD: unused. */ class IAudioRecordCallback : public virtual RefBase { friend AudioRecord; protected: // Request for client to read newly available data. // Used for TRANSFER_CALLBACK mode. // Parameters: // - buffer : Buffer to read from // Returns: // - Number of bytes actually consumed. virtual size_t onMoreData([[maybe_unused]] const AudioRecord::Buffer& buffer) { return 0; } // A buffer overrun occurred. virtual void onOverrun() {} // Record head is at the specified marker (see setMarkerPosition()). virtual void onMarker([[maybe_unused]] uint32_t markerPosition) {} // Record head is at a new position (see setPositionUpdatePeriod()). virtual void onNewPos([[maybe_unused]] uint32_t newPos) {} // IAudioRecord was recreated due to re-routing, server invalidation or // server crash. virtual void onNewIAudioRecord() {} }; /* Returns the minimum frame count required for the successful creation of * an AudioRecord object. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - NO_INIT: audio server or audio hardware not initialized * - BAD_VALUE: unsupported configuration * frameCount is guaranteed to be non-zero if status is NO_ERROR, * and is undefined otherwise. * FIXME This API assumes a route, and so should be deprecated. */ static status_t getMinFrameCount(size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask); /* How data is transferred from AudioRecord */ enum transfer_type { TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters TRANSFER_CALLBACK, // callback EVENT_MORE_DATA TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() TRANSFER_SYNC, // synchronous read() }; /* Constructs an uninitialized AudioRecord. No connection with * AudioFlinger takes place. Use set() after this. * * Parameters: * * client: The attribution source of the owner of the record */ AudioRecord(const android::content::AttributionSourceState& client); /* Creates an AudioRecord object and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. * Unspecified values are set to appropriate default values. * * Parameters: * * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate. * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed * 16 bits per sample). * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. * client: The attribution source of the owner of the record * frameCount: Minimum size of track PCM buffer in frames. This defines the * application's contribution to the * latency of the track. The actual size selected by the AudioRecord could * be larger if the requested size is not compatible with current audio HAL * latency. Zero means to use a default value. * cbf: Callback function. If not null, this function is called periodically * to consume new data in TRANSFER_CALLBACK mode * and inform of marker, position updates, etc. * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames are ready in record track output buffer. * sessionId: Not yet supported. * transferType: How data is transferred from AudioRecord. * flags: See comments on audio_input_flags_t in * pAttributes: If not NULL, supersedes inputSource for use case selection. * threadCanCallJava: Not present in parameter list, and so is fixed at false. */ AudioRecord(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const android::content::AttributionSourceState& client, size_t frameCount = 0, const wp &callback = nullptr, uint32_t notificationFrames = 0, audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, transfer_type transferType = TRANSFER_DEFAULT, audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, const audio_attributes_t* pAttributes = nullptr, audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, audio_microphone_direction_t selectedMicDirection = MIC_DIRECTION_UNSPECIFIED, float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT); /* Terminates the AudioRecord and unregisters it from AudioFlinger. * Also destroys all resources associated with the AudioRecord. */ protected: virtual ~AudioRecord(); public: /* Initialize an AudioRecord that was created using the AudioRecord() constructor. * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. * set() is not multi-thread safe. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful intialization * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized * - PERMISSION_DENIED: recording is not allowed for the requesting process * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. * * Parameters not listed in the AudioRecord constructors above: * * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. */ status_t set(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount = 0, const wp &callback = nullptr, uint32_t notificationFrames = 0, bool threadCanCallJava = false, audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, transfer_type transferType = TRANSFER_DEFAULT, audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, uid_t uid = AUDIO_UID_INVALID, pid_t pid = -1, const audio_attributes_t* pAttributes = nullptr, audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, audio_microphone_direction_t selectedMicDirection = MIC_DIRECTION_UNSPECIFIED, float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT, int32_t maxSharedAudioHistoryMs = 0); /* Result of constructing the AudioRecord. This must be checked for successful initialization * before using any AudioRecord API (except for set()), because using * an uninitialized AudioRecord produces undefined results. * See set() method above for possible return codes. */ status_t initCheck() const { return mStatus; } /* Returns this track's estimated latency in milliseconds. * This includes the latency due to AudioRecord buffer size, resampling if applicable, * and audio hardware driver. */ uint32_t latency() const { return mLatency; } /* getters, see constructor and set() */ audio_format_t format() const { return mFormat; } uint32_t channelCount() const { return mChannelCount; } size_t frameCount() const { return mFrameCount; } size_t frameSize() const { return mFrameSize; } audio_source_t inputSource() const { return mAttributes.source; } audio_channel_mask_t channelMask() const { return mChannelMask; } /* * Return the period of the notification callback in frames. * This value is set when the AudioRecord is constructed. * It can be modified if the AudioRecord is rerouted. */ uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } /* * return metrics information for the current instance. */ status_t getMetrics(mediametrics::Item * &item); /* * Set name of API that is using this object. * For example "aaudio" or "opensles". * This may be logged or reported as part of MediaMetrics. */ void setCallerName(const std::string &name) { mCallerName = name; } std::string getCallerName() const { return mCallerName; }; /* After it's created the track is not active. Call start() to * make it active. If set, the callback will start being called. * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until * the specified event occurs on the specified trigger session. */ status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, audio_session_t triggerSession = AUDIO_SESSION_NONE); /* Stop a track. The callback will cease being called. Note that obtainBuffer() still * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. */ void stop(); bool stopped() const; /* Calls stop() and then wait for all of the callbacks to return. * It is safe to call this if stop() or pause() has already been called. * * This function is called from the destructor. But since AudioRecord * is ref counted, the destructor may be called later than desired. * This can be called explicitly as part of closing an AudioRecord * if you want to be certain that callbacks have completely finished. * * This is not thread safe and should only be called from one thread, * ideally as the AudioRecord is being closed. */ void stopAndJoinCallbacks(); /* Return the sink sample rate for this record track in Hz. * If specified as zero in constructor or set(), this will be the source sample rate. * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. */ uint32_t getSampleRate() const { return mSampleRate; } /* Return the sample rate from the AudioFlinger input thread. */ uint32_t getHalSampleRate() const; /* Return the channel count from the AudioFlinger input thread. */ uint32_t getHalChannelCount() const; /* Return the HAL format from the AudioFlinger input thread. */ audio_format_t getHalFormat() const; /* Sets marker position. When record reaches the number of frames specified, * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition * with marker == 0 cancels marker notification callback. * To set a marker at a position which would compute as 0, * a workaround is to set the marker at a nearby position such as ~0 or 1. * If the AudioRecord has been opened with no callback function associated, * the operation will fail. * * Parameters: * * marker: marker position expressed in wrapping (overflow) frame units, * like the return value of getPosition(). * * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - INVALID_OPERATION: the AudioRecord has no callback installed. */ status_t setMarkerPosition(uint32_t marker); status_t getMarkerPosition(uint32_t *marker) const; /* Sets position update period. Every time the number of frames specified has been recorded, * a callback with event type EVENT_NEW_POS is called. * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification * callback. * If the AudioRecord has been opened with no callback function associated, * the operation will fail. * Extremely small values may be rounded up to a value the implementation can support. * * Parameters: * * updatePeriod: position update notification period expressed in frames. * * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - INVALID_OPERATION: the AudioRecord has no callback installed. */ status_t setPositionUpdatePeriod(uint32_t updatePeriod); status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; /* Return the total number of frames recorded since recording started. * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. * It is reset to zero by stop(). * * Parameters: * * position: Address where to return record head position. * * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - BAD_VALUE: position is NULL */ status_t getPosition(uint32_t *position) const; /* Return the record timestamp. * * Parameters: * timestamp: A pointer to the timestamp to be filled. * * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - BAD_VALUE: timestamp is NULL */ status_t getTimestamp(ExtendedTimestamp *timestamp); /** * @param transferType * @return text string that matches the enum name */ static const char * convertTransferToText(transfer_type transferType); /* Returns a handle on the audio input used by this AudioRecord. * * Parameters: * none. * * Returned value: * handle on audio hardware input */ // FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp audio_io_handle_t getInput() const __attribute__((__deprecated__)) { return getInputPrivate(); } private: audio_io_handle_t getInputPrivate() const; public: /* Returns the audio session ID associated with this AudioRecord. * * Parameters: * none. * * Returned value: * AudioRecord session ID. * * No lock needed because session ID doesn't change after first set(). */ audio_session_t getSessionId() const { return mSessionId; } /* Public API for TRANSFER_OBTAIN mode. * Obtains a buffer of up to "audioBuffer->frameCount" full frames. * After draining these frames of data, the caller should release them with releaseBuffer(). * If the track buffer is not empty, obtainBuffer() returns as many contiguous * full frames as are available immediately. * * If nonContig is non-NULL, it is an output parameter that will be set to the number of * additional non-contiguous frames that are predicted to be available immediately, * if the client were to release the first frames and then call obtainBuffer() again. * This value is only a prediction, and needs to be confirmed. * It will be set to zero for an error return. * * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK * regardless of the value of waitCount. * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a * maximum timeout based on waitCount; see chart below. * Buffers will be returned until the pool * is exhausted, at which point obtainBuffer() will either block * or return WOULD_BLOCK depending on the value of the "waitCount" * parameter. * * Interpretation of waitCount: * +n limits wait time to n * WAIT_PERIOD_MS, * -1 causes an (almost) infinite wait time, * 0 non-blocking. * * Buffer fields * On entry: * frameCount number of frames requested * size ignored * raw ignored * sequence ignored * After error return: * frameCount 0 * size 0 * raw undefined * sequence undefined * After successful return: * frameCount actual number of frames available, <= number requested * size actual number of bytes available * raw pointer to the buffer * sequence IAudioRecord instance sequence number, as of obtainBuffer() */ status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig = NULL); // Explicit Routing /** * TODO Document this method. */ status_t setInputDevice(audio_port_handle_t deviceId); /** * TODO Document this method. */ audio_port_handle_t getInputDevice(); /* Returns the ID of the audio device actually used by the input to which this AudioRecord * is attached. * The device ID is relevant only if the AudioRecord is active. * When the AudioRecord is inactive, the device ID returned can be either: * - AUDIO_PORT_HANDLE_NONE if the AudioRecord is not attached to any output. * - The device ID used before paused or stopped. * - The device ID selected by audio policy manager of setOutputDevice() if the AudioRecord * has not been started yet. * * Parameters: * none. */ audio_port_handle_t getRoutedDeviceId(); /* Add an AudioDeviceCallback. The caller will be notified when the audio device * to which this AudioRecord is routed is updated. * Replaces any previously installed callback. * Parameters: * callback: The callback interface * Returns NO_ERROR if successful. * INVALID_OPERATION if the same callback is already installed. * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable * BAD_VALUE if the callback is NULL */ status_t addAudioDeviceCallback( const sp& callback); /* remove an AudioDeviceCallback. * Parameters: * callback: The callback interface * Returns NO_ERROR if successful. * INVALID_OPERATION if the callback is not installed * BAD_VALUE if the callback is NULL */ status_t removeAudioDeviceCallback( const sp& callback); // AudioSystem::AudioDeviceCallback> virtuals virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, audio_port_handle_t deviceId); private: /* If nonContig is non-NULL, it is an output parameter that will be set to the number of * additional non-contiguous frames that are predicted to be available immediately, * if the client were to release the first frames and then call obtainBuffer() again. * This value is only a prediction, and needs to be confirmed. * It will be set to zero for an error return. * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), * in case the requested amount of frames is in two or more non-contiguous regions. * FIXME requested and elapsed are both relative times. Consider changing to absolute time. */ status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, struct timespec *elapsed = NULL, size_t *nonContig = NULL); public: /* Public API for TRANSFER_OBTAIN mode. * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. * * Buffer fields: * frameCount currently ignored but recommend to set to actual number of frames consumed * size actual number of bytes consumed, must be multiple of frameSize * raw ignored */ void releaseBuffer(const Buffer* audioBuffer); /* As a convenience we provide a read() interface to the audio buffer. * Input parameter 'size' is in byte units. * This is implemented on top of obtainBuffer/releaseBuffer. For best * performance use callbacks. Returns actual number of bytes read >= 0, * or one of the following negative status codes: * INVALID_OPERATION AudioRecord is configured for streaming mode * BAD_VALUE size is invalid * WOULD_BLOCK when obtainBuffer() returns same, or * AudioRecord was stopped during the read * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). * Default behavior is to only return when all data has been transferred. Set 'blocking' to * false for the method to return immediately without waiting to try multiple times to read * the full content of the buffer. */ ssize_t read(void* buffer, size_t size, bool blocking = true); /* Return the number of input frames lost in the audio driver since the last call of this * function. Audio driver is expected to reset the value to 0 and restart counting upon * returning the current value by this function call. Such loss typically occurs when the * user space process is blocked longer than the capacity of audio driver buffers. * Units: the number of input audio frames. * FIXME The side-effect of resetting the counter may be incompatible with multi-client. * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. */ uint32_t getInputFramesLost() const; /* Get the flags */ audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } /* Get active microphones. A empty vector of MicrophoneInfoFw will be passed as a parameter, * the data will be filled when querying the hal. */ status_t getActiveMicrophones( std::vector* activeMicrophones); /* Set the Microphone direction (for processing purposes). */ status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction); /* Set the Microphone zoom factor (for processing purposes). */ status_t setPreferredMicrophoneFieldDimension(float zoom); /* Get the unique port ID assigned to this AudioRecord instance by audio policy manager. * The ID is unique across all audioserver clients and can change during the life cycle * of a given AudioRecord instance if the connection to audioserver is restored. */ audio_port_handle_t getPortId() const { return mPortId; }; /* Sets the LogSessionId field which is used for metrics association of * this object with other objects. A nullptr or empty string clears * the logSessionId. */ void setLogSessionId(const char *logSessionId); status_t shareAudioHistory(const std::string& sharedPackageName, int64_t sharedStartMs); /* * Dumps the state of an audio record. */ status_t dump(int fd, const Vector& args) const; private: /* copying audio record objects is not allowed */ AudioRecord(const AudioRecord& other); AudioRecord& operator = (const AudioRecord& other); /* a small internal class to handle the callback */ class AudioRecordThread : public Thread { public: AudioRecordThread(AudioRecord& receiver); // Do not call Thread::requestExitAndWait() without first calling requestExit(). // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. virtual void requestExit(); void pause(); // suspend thread from execution at next loop boundary void resume(); // allow thread to execute, if not requested to exit void wake(); // wake to handle changed notification conditions. private: void pauseInternal(nsecs_t ns = 0LL); // like pause(), but only used internally within thread friend class AudioRecord; virtual bool threadLoop(); AudioRecord& mReceiver; virtual ~AudioRecordThread(); Mutex mMyLock; // Thread::mLock is private Condition mMyCond; // Thread::mThreadExitedCondition is private bool mPaused; // whether thread is requested to pause at next loop entry bool mPausedInt; // whether thread internally requests pause nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored bool mIgnoreNextPausedInt; // skip any internal pause and go immediately // to processAudioBuffer() as state may have changed // since pause time calculated. }; // body of AudioRecordThread::threadLoop() // returns the maximum amount of time before we would like to run again, where: // 0 immediately // > 0 no later than this many nanoseconds from now // NS_WHENEVER still active but no particular deadline // NS_INACTIVE inactive so don't run again until re-started // NS_NEVER never again static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; nsecs_t processAudioBuffer(); // caller must hold lock on mLock for all _l methods status_t createRecord_l(const Modulo &epoch); // FIXME enum is faster than strcmp() for parameter 'from' status_t restoreRecord_l(const char *from); void updateRoutedDeviceId_l(); sp mAudioRecordThread; mutable Mutex mLock; std::unique_ptr mTracker; // Current client state: false = stopped, true = active. Protected by mLock. If more states // are added, consider changing this to enum State { ... } mState as in AudioTrack. bool mActive = false; // for client callback handler wp mCallback; sp mLegacyCallbackWrapper; bool mInitialized = false; // Protect against double set // for notification APIs uint32_t mNotificationFramesReq; // requested number of frames between each // notification callback // as specified in constructor or set() uint32_t mNotificationFramesAct; // actual number of frames between each // notification callback bool mRefreshRemaining; // processAudioBuffer() should refresh // mRemainingFrames and mRetryOnPartialBuffer // These are private to processAudioBuffer(), and are not protected by a lock uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() uint32_t mObservedSequence; // last observed value of mSequence Modulo mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; Modulo mNewPosition; // in frames uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS status_t mStatus = NO_INIT; android::content::AttributionSourceState mClientAttributionSource; // Owner's attribution source size_t mFrameCount; // corresponds to current IAudioRecord, value is // reported back by AudioFlinger to the client size_t mReqFrameCount; // frame count to request the first or next time // a new IAudioRecord is needed, non-decreasing int64_t mFramesRead; // total frames read. reset to zero after // the start() following stop(). It is not // changed after restoring the track. int64_t mFramesReadServerOffset; // An offset to server frames read due to // restoring AudioRecord, or stop/start. // constant after constructor or set() uint32_t mSampleRate; audio_format_t mFormat; uint32_t mChannelCount; size_t mFrameSize; // app-level frame size == AudioFlinger frame size uint32_t mLatency; // in ms audio_channel_mask_t mChannelMask; audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may // be denied by client or server, such as // AUDIO_INPUT_FLAG_FAST. mLock must be // held to read or write those bits reliably. audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const audio_session_t mSessionId = AUDIO_SESSION_ALLOCATE; audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE; /** * mLogSessionId is a string identifying this AudioRecord for the metrics service. * It may be unique or shared with other objects. An empty string means the * logSessionId is not set. */ std::string mLogSessionId{}; transfer_type mTransfer; // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 // provided the initial set() was successful sp mAudioRecord; sp mCblkMemory; audio_track_cblk_t* mCblk; // re-load after mLock.unlock() sp mBufferMemory; audio_io_handle_t mInput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getInputforAttr() int mPreviousPriority = ANDROID_PRIORITY_NORMAL; // before start() SchedPolicy mPreviousSchedulingGroup = SP_DEFAULT; bool mAwaitBoost = false; // thread should wait for priority boost before running // The proxy should only be referenced while a lock is held because the proxy isn't // multi-thread safe. // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, // provided that the caller also holds an extra reference to the proxy and shared memory to keep // them around in case they are replaced during the obtainBuffer(). sp mProxy; bool mInOverrun; // whether recorder is currently in overrun state ExtendedTimestamp mPreviousTimestamp{}; // used to detect retrograde motion bool mTimestampRetrogradePositionReported = false; // reduce log spam bool mTimestampRetrogradeTimeReported = false; // reduce log spam // Format conversion. Maybe needed for adding fast tracks whose format is different from server. audio_config_base_t mServerConfig; size_t mServerFrameSize; size_t mServerSampleSize; std::unique_ptr mFormatConversionBufRaw; Buffer mFormatConversionBuffer; uint32_t mHalSampleRate; // AudioFlinger thread sample rate uint32_t mHalChannelCount; // AudioFlinger thread channel count audio_format_t mHalFormat; // AudioFlinger thread format private: class DeathNotifier : public IBinder::DeathRecipient { public: DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } protected: virtual void binderDied(const wp& who); private: const wp mAudioRecord; }; sp mDeathNotifier; uint32_t mSequence; // incremented for each new IAudioRecord attempt audio_attributes_t mAttributes; // For Device Selection API // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. // Device requested by the application. audio_port_handle_t mSelectedDeviceId = AUDIO_PORT_HANDLE_NONE; // Device actually selected by AudioPolicyManager: This may not match the app // selection depending on other activity and connected devices audio_port_handle_t mRoutedDeviceId = AUDIO_PORT_HANDLE_NONE; wp mDeviceCallback; audio_microphone_direction_t mSelectedMicDirection = MIC_DIRECTION_UNSPECIFIED; float mSelectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT; int32_t mMaxSharedAudioHistoryMs = 0; std::string mSharedAudioPackageName = {}; int64_t mSharedAudioStartMs = 0; private: class MediaMetrics { public: MediaMetrics() : mMetricsItem(mediametrics::Item::create("audiorecord")), mCreatedNs(systemTime(SYSTEM_TIME_REALTIME)), mStartedNs(0), mDurationNs(0), mCount(0), mLastError(NO_ERROR) { } ~MediaMetrics() { // mMetricsItem alloc failure will be flagged in the constructor // don't log empty records if (mMetricsItem->count() > 0) { mMetricsItem->selfrecord(); } } void gather(const AudioRecord *record); mediametrics::Item *dup() { return mMetricsItem->dup(); } void logStart(nsecs_t when) { mStartedNs = when; mCount++; } void logStop(nsecs_t when) { mDurationNs += (when-mStartedNs); mStartedNs = 0;} void markError(status_t errcode, const char *func) { mLastError = errcode; mLastErrorFunc = func;} private: std::unique_ptr mMetricsItem; nsecs_t mCreatedNs; // XXX: perhaps not worth it in production nsecs_t mStartedNs; nsecs_t mDurationNs; int32_t mCount; status_t mLastError; std::string mLastErrorFunc; }; MediaMetrics mMediaMetrics; std::string mMetricsId; // GUARDED_BY(mLock), could change in createRecord_l(). std::string mCallerName; // for example "aaudio" void reportError(status_t status, const char *event, const char *message) const; }; }; // namespace android #endif // ANDROID_AUDIORECORD_H