| /* |
| * 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 <cutils/sched_policy.h> |
| #include <media/AudioSystem.h> |
| #include <media/AudioTimestamp.h> |
| #include <media/IAudioRecord.h> |
| #include <media/Modulo.h> |
| #include <utils/threads.h> |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| |
| struct audio_track_cblk_t; |
| class AudioRecordClientProxy; |
| |
| // ---------------------------------------------------------------------------- |
| |
| class AudioRecord : public AudioSystem::AudioDeviceCallback |
| { |
| public: |
| |
| /* Events used by AudioRecord callback function (callback_t). |
| * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*. |
| */ |
| enum event_type { |
| EVENT_MORE_DATA = 0, // Request to read available data from buffer. |
| // If this event is delivered but the callback handler |
| // does not want to read the available data, the handler must |
| // explicitly ignore the event by setting frameCount to zero. |
| EVENT_OVERRUN = 1, // Buffer overrun occurred. |
| EVENT_MARKER = 2, // Record head is at the specified marker position |
| // (See setMarkerPosition()). |
| EVENT_NEW_POS = 3, // Record head is at a new position |
| // (See setPositionUpdatePeriod()). |
| EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and |
| // voluntary invalidation by mediaserver, or mediaserver crash. |
| }; |
| |
| /* Client should declare a Buffer and pass address to obtainBuffer() |
| * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. |
| */ |
| |
| class Buffer |
| { |
| public: |
| // FIXME use m prefix |
| 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 |
| |
| size_t size; // 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; |
| short* i16; // signed 16-bit |
| int8_t* i8; // unsigned 8-bit, offset by 0x80 |
| // input to obtainBuffer(): unused, output: pointer to buffer |
| }; |
| }; |
| |
| /* 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. |
| */ |
| |
| typedef void (*callback_t)(int event, void* user, void *info); |
| |
| /* 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: |
| * |
| * opPackageName: The package name used for app ops. |
| */ |
| AudioRecord(const String16& opPackageName); |
| |
| /* 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. |
| * opPackageName: The package name used for app ops. |
| * 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 <system/audio.h> |
| * 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 String16& opPackageName, |
| size_t frameCount = 0, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| 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, |
| uid_t uid = AUDIO_UID_INVALID, |
| pid_t pid = -1, |
| const audio_attributes_t* pAttributes = NULL); |
| |
| /* 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, |
| callback_t cbf = NULL, |
| void* user = NULL, |
| 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 = NULL); |
| |
| /* 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; } |
| |
| /* |
| * 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; } |
| |
| /* 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; |
| |
| /* 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; } |
| |
| /* 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 |
| * After error return: |
| * frameCount 0 |
| * size 0 |
| * raw undefined |
| * After successful return: |
| * frameCount actual number of frames available, <= number requested |
| * size actual number of bytes available |
| * raw pointer to the buffer |
| */ |
| |
| 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<AudioSystem::AudioDeviceCallback>& 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<AudioSystem::AudioDeviceCallback>& 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; } |
| |
| 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, bool bCanCallJava = false); |
| |
| // 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 openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName); |
| |
| // FIXME enum is faster than strcmp() for parameter 'from' |
| status_t restoreRecord_l(const char *from); |
| |
| void updateRoutedDeviceId_l(); |
| |
| sp<AudioRecordThread> mAudioRecordThread; |
| mutable Mutex mLock; |
| |
| // 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; |
| |
| // for client callback handler |
| callback_t mCbf; // callback handler for events, or NULL |
| void* mUserData; |
| |
| // 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<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units |
| bool mMarkerReached; |
| Modulo<uint32_t> mNewPosition; // in frames |
| uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS |
| |
| status_t mStatus; |
| |
| String16 mOpPackageName; // The package name used for app ops. |
| |
| 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; |
| transfer_type mTransfer; |
| |
| // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 |
| // provided the initial set() was successful |
| sp<IAudioRecord> mAudioRecord; |
| sp<IMemory> mCblkMemory; |
| audio_track_cblk_t* mCblk; // re-load after mLock.unlock() |
| sp<IMemory> mBufferMemory; |
| audio_io_handle_t mInput; // returned by AudioSystem::getInput() |
| |
| int mPreviousPriority; // before start() |
| SchedPolicy mPreviousSchedulingGroup; |
| bool mAwaitBoost; // 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<AudioRecordClientProxy> mProxy; |
| |
| bool mInOverrun; // whether recorder is currently in overrun state |
| |
| private: |
| class DeathNotifier : public IBinder::DeathRecipient { |
| public: |
| DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } |
| protected: |
| virtual void binderDied(const wp<IBinder>& who); |
| private: |
| const wp<AudioRecord> mAudioRecord; |
| }; |
| |
| sp<DeathNotifier> mDeathNotifier; |
| uint32_t mSequence; // incremented for each new IAudioRecord attempt |
| uid_t mClientUid; |
| pid_t mClientPid; |
| audio_attributes_t mAttributes; |
| |
| // For Device Selection API |
| // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. |
| audio_port_handle_t mSelectedDeviceId; // Device requested by the application. |
| audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: |
| // May not match the app selection depending on other |
| // activity and connected devices |
| wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; |
| audio_port_handle_t mPortId; // unique ID allocated by audio policy |
| |
| }; |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIORECORD_H |
