/* * 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 AUDIORECORD_H_ #define AUDIORECORD_H_ #include #include #include #include #include #include #include #include namespace android { class audio_track_cblk_t; // ---------------------------------------------------------------------------- class AudioRecord : virtual public RefBase { public: static const int DEFAULT_SAMPLE_RATE = 8000; /* 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 more data from PCM buffer. EVENT_OVERRUN = 1, // PCM buffer overrun occured. 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()). }; /* Create Buffer on the stack and pass it to obtainBuffer() * and releaseBuffer(). */ class Buffer { public: enum { MUTE = 0x00000001 }; uint32_t flags; int channelCount; audio_format_t format; size_t frameCount; size_t size; // total size in bytes == frameCount * frameSize union { void* raw; short* i16; int8_t* i8; }; }; /* 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 ready or an overrun condition occurs. * 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 less bytes are * read. * - 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. */ 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 */ static status_t getMinFrameCount(int* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask); /* Constructs an uninitialized AudioRecord. No connection with * AudioFlinger takes place. */ AudioRecord(); /* Creates an AudioRecord track and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. * Unspecified values are set to the audio hardware's current * values. * * Parameters: * * inputSource: Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT). * sampleRate: Track sampling rate in Hz. * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed * 16 bits per sample). * channelMask: Channel mask. * frameCount: Total size of track PCM buffer in frames. This defines the * latency of the track. * cbf: Callback function. If not null, this function is called periodically * to provide new PCM data. * 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. */ AudioRecord(audio_source_t inputSource, uint32_t sampleRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO, int frameCount = 0, callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, int sessionId = 0); /* Terminates the AudioRecord and unregisters it from AudioFlinger. * Also destroys all resources associated with the AudioRecord. */ ~AudioRecord(); /* Initialize an uninitialized AudioRecord. * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful intialization * - INVALID_OPERATION: AudioRecord is already intitialized or record device is already in use * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized * - PERMISSION_DENIED: recording is not allowed for the requesting process * */ status_t set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT, uint32_t sampleRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO, int frameCount = 0, callback_t cbf = NULL, void* user = NULL, int notificationFrames = 0, bool threadCanCallJava = false, int sessionId = 0); /* Result of constructing the AudioRecord. This must be checked * before using any AudioRecord API (except for set()), using * an uninitialized AudioRecord produces undefined results. * See set() method above for possible return codes. */ status_t initCheck() const; /* Returns this track's latency in milliseconds. * This includes the latency due to AudioRecord buffer size * and audio hardware driver. */ uint32_t latency() const; /* getters, see constructor and set() */ audio_format_t format() const; int channelCount() const; uint32_t frameCount() const; size_t frameSize() const; audio_source_t inputSource() const; /* 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, int triggerSession = 0); /* Stop a track. If set, the callback will cease being called and * obtainBuffer returns STOPPED. Note that obtainBuffer() still works * and will fill up buffers until the pool is exhausted. */ void stop(); bool stopped() const; /* get sample rate for this record track */ uint32_t getSampleRate() 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. * If the AudioRecord has been opened with no callback function associated, * the operation will fail. * * Parameters: * * marker: marker position 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 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. * * 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; /* Gets record head position. The position is the total number of frames * recorded since record start. * * Parameters: * * position: Address where to return record head position within AudioRecord buffer. * * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - BAD_VALUE: position is NULL */ status_t getPosition(uint32_t *position) const; /* returns a handle on the audio input used by this AudioRecord. * * Parameters: * none. * * Returned value: * handle on audio hardware input */ audio_io_handle_t getInput() const; /* returns the audio session ID associated with this AudioRecord. * * Parameters: * none. * * Returned value: * AudioRecord session ID. */ int getSessionId() const; /* obtains a buffer of "frameCount" frames. The buffer must be * filled entirely. If the track is stopped, obtainBuffer() returns * STOPPED instead of NO_ERROR as long as there are buffers available, * at which point NO_MORE_BUFFERS is returned. * Buffers will be returned until the pool (buffercount()) * is exhausted, at which point obtainBuffer() will either block * or return WOULD_BLOCK depending on the value of the "blocking" * parameter. */ enum { NO_MORE_BUFFERS = 0x80000001, STOPPED = 1 }; status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); void releaseBuffer(Buffer* audioBuffer); /* As a convenience we provide a read() interface to the audio buffer. * This is implemented on top of obtainBuffer/releaseBuffer. */ ssize_t read(void* buffer, size_t size); /* Return the amount 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. * Unit: the number of input audio frames */ unsigned int getInputFramesLost() const; private: /* copying audio tracks 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 private: 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 currently paused }; // body of AudioRecordThread::threadLoop() bool processAudioBuffer(const sp& thread); status_t openRecord_l(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_io_handle_t input); audio_io_handle_t getInput_l(); status_t restoreRecord_l(audio_track_cblk_t*& cblk); sp mAudioRecordThread; mutable Mutex mLock; bool mActive; // protected by mLock // for client callback handler callback_t mCbf; void* mUserData; // for notification APIs uint32_t mNotificationFrames; uint32_t mRemainingFrames; uint32_t mMarkerPosition; // in frames bool mMarkerReached; uint32_t mNewPosition; // in frames uint32_t mUpdatePeriod; // in ms // constant after constructor or set() uint32_t mFrameCount; audio_format_t mFormat; uint8_t mChannelCount; audio_source_t mInputSource; status_t mStatus; uint32_t mLatency; audio_channel_mask_t mChannelMask; audio_io_handle_t mInput; // returned by AudioSystem::getInput() int mSessionId; // may be changed if IAudioRecord object is re-created sp mAudioRecord; sp mCblkMemory; audio_track_cblk_t* mCblk; int mPreviousPriority; // before start() SchedPolicy mPreviousSchedulingGroup; }; }; // namespace android #endif /*AUDIORECORD_H_*/