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M7350/base/include/media/AudioRecord.h
T f9cc65cfda M7350v1_en_gpl
2024-09-09 08:52:07 +00:00

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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
*
* 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 <stdint.h>
#include <sys/types.h>
#include <media/IAudioFlinger.h>
#include <media/IAudioRecord.h>
#include <media/AudioTrack.h>
#include <utils/RefBase.h>
#include <utils/Errors.h>
#include <binder/IInterface.h>
#include <binder/IMemory.h>
#include <utils/threads.h>
namespace android {
// ----------------------------------------------------------------------------
class AudioRecord
{
public:
static const int DEFAULT_SAMPLE_RATE = 8000;
/* Events used by AudioRecord callback function (callback_t).
*
* to keep in sync with frameworks/base/media/java/android/media/AudioRecord.java
*/
enum event_type {
EVENT_MORE_DATA = 0, // Request to reqd 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;
int format;
size_t frameCount;
size_t size;
union {
void* raw;
short* i16;
int8_t* i8;
};
};
/* These are static methods to control the system-wide AudioFlinger
* only privileged processes can have access to them
*/
// static status_t setMasterMute(bool mute);
/* 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 an uin32_t containing the marker position in frames.
* - EVENT_NEW_POS: pointer to an uin32_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,
int format,
int channelCount);
/* 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 AudioSystem::PCM_16_BIT for signed
* 16 bits per sample).
* channels: Channel mask: see AudioSystem::audio_channels.
* frameCount: Total size of track PCM buffer in frames. This defines the
* latency of the track.
* flags: A bitmask of acoustic values from enum record_flags. It enables
* AGC, NS, and IIR.
* cbf: Callback function. If not null, this function is called periodically
* to provide new PCM data.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames are ready in record track output buffer.
* user Context for use by the callback receiver.
*/
enum record_flags {
RECORD_AGC_ENABLE = 1,
RECORD_NS_ENABLE = 2,
RECORD_IIR_ENABLE = 4
};
AudioRecord(int inputSource,
uint32_t sampleRate = 0,
int format = 0,
uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
int sessionId = 0);
/* Terminates the AudioRecord and unregisters it from AudioFlinger.
* Also destroys all resources assotiated 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(int inputSource = 0,
uint32_t sampleRate = 0,
int format = 0,
uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
void* user = 0,
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 */
int format() const;
int channelCount() const;
int channels() const;
uint32_t frameCount() const;
int frameSize() const;
int 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.
*/
status_t start();
/* 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.
*/
status_t stop();
bool stopped() const;
/* get sample rate for this record track
*/
uint32_t getSampleRate();
/* 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);
/* 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);
/* 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);
/* 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();
/* returns the audio session ID associated to this AudioRecord.
*
* Parameters:
* none.
*
* Returned value:
* AudioRecord session ID.
*/
int getSessionId();
/* 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 availlable,
* 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 lockBuffer/unlockBuffer.
*/
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();
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 ClientRecordThread : public Thread
{
public:
ClientRecordThread(AudioRecord& receiver, bool bCanCallJava = false);
private:
friend class AudioRecord;
virtual bool threadLoop();
virtual status_t readyToRun() { return NO_ERROR; }
virtual void onFirstRef() {}
AudioRecord& mReceiver;
Mutex mLock;
};
bool processAudioBuffer(const sp<ClientRecordThread>& thread);
status_t openRecord(uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
audio_io_handle_t input);
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
sp<ClientRecordThread> mClientRecordThread;
Mutex mRecordThreadLock;
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
int32_t mFormat;
uint8_t mChannelCount;
uint8_t mInputSource;
uint8_t mReserved;
status_t mStatus;
uint32_t mLatency;
volatile int32_t mActive;
callback_t mCbf;
void* mUserData;
uint32_t mNotificationFrames;
uint32_t mRemainingFrames;
uint32_t mMarkerPosition;
bool mMarkerReached;
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
uint32_t mFlags;
uint32_t mChannels;
audio_io_handle_t mInput;
bool mFirstread;
int mSessionId;
};
}; // namespace android
#endif /*AUDIORECORD_H_*/
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