| /* |
| * 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 <binder/IMemory.h> |
| #include <cutils/sched_policy.h> |
| #include <media/AudioSystem.h> |
| #include <media/IAudioRecord.h> |
| #include <system/audio.h> |
| #include <utils/RefBase.h> |
| #include <utils/Errors.h> |
| #include <utils/threads.h> |
| |
| namespace android { |
| |
| class audio_track_cblk_t; |
| class AudioRecordClientProxy; |
| |
| // ---------------------------------------------------------------------------- |
| |
| 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 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()). |
| }; |
| |
| /* Client should declare Buffer on the stack and pass address to obtainBuffer() |
| * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. |
| */ |
| |
| class Buffer |
| { |
| public: |
| size_t frameCount; // number of sample frames corresponding to size; |
| // on input it is the number of frames available, |
| // on output is the number of frames actually drained |
| |
| size_t size; // total size in bytes == frameCount * frameSize |
| union { |
| void* raw; |
| short* i16; // signed 16-bit |
| int8_t* i8; // unsigned 8-bit, offset by 0x80 |
| }; |
| }; |
| |
| /* 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 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. |
| */ |
| |
| 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(size_t* 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 object 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: 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 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()), because using |
| * an uninitialized AudioRecord produces undefined results. |
| * See set() method above for possible return codes. |
| */ |
| status_t initCheck() const; |
| |
| /* Returns this track's estimated 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; |
| uint32_t channelCount() const; |
| size_t frameCount() const; |
| size_t frameSize() const { return mFrameSize; } |
| 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 drain buffers until the pool is exhausted. |
| */ |
| void stop(); |
| bool stopped() const; |
| |
| /* Get sample rate for this record track in Hz. |
| */ |
| 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 |
| * drained entirely, and then released with releaseBuffer(). |
| * 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 |
| * is exhausted, at which point obtainBuffer() will either block |
| * or return WOULD_BLOCK depending on the value of the "blocking" |
| * parameter. |
| * |
| * Interpretation of waitCount: |
| * +n limits wait time to n * WAIT_PERIOD_MS, |
| * -1 causes an (almost) infinite wait time, |
| * 0 non-blocking. |
| */ |
| |
| enum { |
| NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value |
| STOPPED = 1 |
| }; |
| |
| status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); |
| |
| /* Release an emptied buffer of "frameCount" frames for AudioFlinger to re-fill. */ |
| 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 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. |
| */ |
| unsigned int getInputFramesLost() 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, 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<AudioRecordThread>& thread); |
| |
| status_t openRecord_l(uint32_t sampleRate, |
| audio_format_t format, |
| size_t frameCount, |
| audio_io_handle_t input); |
| audio_io_handle_t getInput_l(); |
| status_t restoreRecord_l(audio_track_cblk_t*& cblk); |
| |
| sp<AudioRecordThread> mAudioRecordThread; |
| mutable Mutex mLock; |
| |
| bool mActive; // protected by mLock |
| |
| // for client callback handler |
| callback_t mCbf; // callback handler for events, or NULL |
| 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 mSampleRate; |
| size_t mFrameCount; |
| audio_format_t mFormat; |
| uint8_t mChannelCount; |
| size_t mFrameSize; // app-level frame size == AudioFlinger frame size |
| 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<IAudioRecord> mAudioRecord; |
| sp<IMemory> mCblkMemory; |
| audio_track_cblk_t* mCblk; |
| void* mBuffers; // starting address of buffers in shared memory |
| |
| int mPreviousPriority; // before start() |
| SchedPolicy mPreviousSchedulingGroup; |
| AudioRecordClientProxy* mProxy; |
| }; |
| |
| }; // namespace android |
| |
| #endif /*AUDIORECORD_H_*/ |
