/*
* 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_*/