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* 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
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* See the License for the specific language governing permissions and
* limitations under the License.
#include <cutils/sched_policy.h>
#include <media/AudioSystem.h>
#include <media/IAudioRecord.h>
#include <utils/threads.h>
namespace android {
// ----------------------------------------------------------------------------
class audio_track_cblk_t;
class AudioRecordClientProxy;
// ----------------------------------------------------------------------------
class AudioRecord : public RefBase
/* Events used by AudioRecord callback function (callback_t).
* Keep in sync with frameworks/base/media/java/android/media/ 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()).
EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and
// voluntary invalidation by mediaserver, or mediaserver crash.
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
* and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
class Buffer
// FIXME use m prefix
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
// (currently ignored, but will make the primary field in future)
size_t size; // input/output in bytes == frameCount * frameSize
// FIXME this is redundant with respect to frameCount,
// and TRANSFER_OBTAIN mode is broken for 8-bit data
// since we don't define the frame format
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);
/* How data is transferred from AudioRecord
enum transfer_type {
TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters
TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
TRANSFER_SYNC, // synchronous read()
/* Constructs an uninitialized AudioRecord. No connection with
* AudioFlinger takes place. Use set() after this.
/* 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.
* 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.
* 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 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>
* 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,
int frameCount = 0,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
int sessionId = 0,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
/* Terminates the AudioRecord and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioRecord.
virtual ~AudioRecord();
/* 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.
* 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 (channels, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
* 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,
int frameCount = 0,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
bool threadCanCallJava = false,
int sessionId = 0,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
/* 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 { return mStatus; }
/* 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 { 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 mInputSource; }
/* 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. 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.
* 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 the 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;
/* 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.
* No lock needed because session ID doesn't change after first set().
int getSessionId() const { return mSessionId; }
/* 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 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.
* obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
* which should use read() or callback EVENT_MORE_DATA instead.
* 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
* 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
/* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
/* If nonContig is non-NULL, it is an output parameter that will be set to the number of
* additional non-contiguous frames that are available immediately.
* 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);
/* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */
// FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
void releaseBuffer(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().
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;
/* 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
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 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
// 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(const sp<AudioRecordThread>& thread);
// caller must hold lock on mLock for all _l methods
status_t openRecord_l(size_t epoch);
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreRecord_l(const char *from);
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
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback
bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
// 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()
int mObservedSequence; // last observed value of mSequence
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
uint32_t mNewPosition; // in frames
uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
status_t mStatus;
// constant after constructor or set()
uint32_t mSampleRate;
size_t mFrameCount;
audio_format_t mFormat;
uint32_t mChannelCount;
size_t mFrameSize; // app-level frame size == AudioFlinger frame size
audio_source_t mInputSource;
uint32_t mLatency; // in ms
audio_channel_mask_t mChannelMask;
audio_input_flags_t mFlags;
int mSessionId;
transfer_type mTransfer;
audio_io_handle_t mInput; // returned by AudioSystem::getInput()
// may be changed if IAudioRecord object is re-created
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
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
class DeathNotifier : public IBinder::DeathRecipient {
DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { }
virtual void binderDied(const wp<IBinder>& who);
const wp<AudioRecord> mAudioRecord;
sp<DeathNotifier> mDeathNotifier;
uint32_t mSequence; // incremented for each new IAudioRecord attempt
}; // namespace android