/*
* 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 ANDROID_AUDIORECORD_H
#define ANDROID_AUDIORECORD_H
#include <cutils/sched_policy.h>
#include <media/AudioSystem.h>
#include <media/AudioTimestamp.h>
#include <media/IAudioRecord.h>
#include <media/Modulo.h>
#include <utils/threads.h>
namespace android {
// ----------------------------------------------------------------------------
struct audio_track_cblk_t;
class AudioRecordClientProxy;
// ----------------------------------------------------------------------------
class AudioRecord : public RefBase
{
public:
/* 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 available data from buffer.
// If this event is delivered but the callback handler
// does not want to read the available data, the handler must
// explicitly ignore the event by setting frameCount to zero.
EVENT_OVERRUN = 1, // 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 a Buffer and pass address to obtainBuffer()
* and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
class Buffer
{
public:
// FIXME use m prefix
size_t frameCount; // number of sample frames corresponding to size;
// on input to obtainBuffer() it is the number of frames desired
// on output from obtainBuffer() it is the number of available
// frames to be read
// on input to releaseBuffer() it is currently ignored
size_t size; // input/output in bytes == frameCount * frameSize
// on input to obtainBuffer() it is ignored
// on output from obtainBuffer() it is the number of available
// bytes to be read, which is frameCount * frameSize
// on input to releaseBuffer() it is the number of bytes to
// release
// FIXME This is redundant with respect to frameCount. Consider
// removing size and making frameCount the primary field.
union {
void* raw;
short* i16; // signed 16-bit
int8_t* i8; // unsigned 8-bit, offset by 0x80
// input to obtainBuffer(): unused, output: pointer to buffer
};
};
/* 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 available 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.
* - EVENT_NEW_IAUDIORECORD: unused.
*/
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
* frameCount is guaranteed to be non-zero if status is NO_ERROR,
* and is undefined otherwise.
* FIXME This API assumes a route, and so should be deprecated.
*/
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 the other parameters
TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer()
TRANSFER_SYNC, // synchronous read()
};
/* Constructs an uninitialized AudioRecord. No connection with
* AudioFlinger takes place. Use set() after this.
*
* Parameters:
*
* opPackageName: The package name used for app ops.
*/
AudioRecord(const String16& opPackageName);
/* 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. Zero means to use the source sample rate.
* 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.
* opPackageName: The package name used for app ops.
* 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 data in TRANSFER_CALLBACK mode
* 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>
* pAttributes: If not NULL, supersedes inputSource for use case selection.
* 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,
const String16& opPackageName,
size_t frameCount = 0,
callback_t cbf = NULL,
void* user = NULL,
uint32_t notificationFrames = 0,
audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
uid_t uid = AUDIO_UID_INVALID,
pid_t pid = -1,
const audio_attributes_t* pAttributes = NULL);
/* Terminates the AudioRecord and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioRecord.
*/
protected:
virtual ~AudioRecord();
public:
/* 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.
* set() is not multi-thread safe.
* 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 (channelMask, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
* If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
*
* 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,
size_t frameCount = 0,
callback_t cbf = NULL,
void* user = NULL,
uint32_t notificationFrames = 0,
bool threadCanCallJava = false,
audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
uid_t uid = AUDIO_UID_INVALID,
pid_t pid = -1,
const audio_attributes_t* pAttributes = NULL);
/* Result of constructing the AudioRecord. This must be checked for successful initialization
* 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, resampling if applicable,
* 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 mAttributes.source; }
/*
* Return the period of the notification callback in frames.
* This value is set when the AudioRecord is constructed.
* It can be modified if the AudioRecord is rerouted.
*/
uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; }
/* 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,
audio_session_t triggerSession = AUDIO_SESSION_NONE);
/* Stop a track. 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.
* If specified as zero in constructor or set(), this will be the source sample rate.
* 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 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;
/* Return the record timestamp.
*
* Parameters:
* timestamp: A pointer to the timestamp to be filled.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - BAD_VALUE: timestamp is NULL
*/
status_t getTimestamp(ExtendedTimestamp *timestamp);
/* Returns a handle on the audio input used by this AudioRecord.
*
* Parameters:
* none.
*
* Returned value:
* handle on audio hardware input
*/
// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp
audio_io_handle_t getInput() const __attribute__((__deprecated__))
{ return getInputPrivate(); }
private:
audio_io_handle_t getInputPrivate() const;
public:
/* 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().
*/
audio_session_t getSessionId() const { return mSessionId; }
/* Public API for TRANSFER_OBTAIN mode.
* 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 nonContig is non-NULL, it is an output parameter that will be set to the number of
* additional non-contiguous frames that are predicted to be available immediately,
* if the client were to release the first frames and then call obtainBuffer() again.
* This value is only a prediction, and needs to be confirmed.
* It will be set to zero for an error return.
*
* 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.
*
* 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
* size ignored
* raw ignored
* 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
*/
status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount,
size_t *nonContig = NULL);
// Explicit Routing
/**
* TODO Document this method.
*/
status_t setInputDevice(audio_port_handle_t deviceId);
/**
* TODO Document this method.
*/
audio_port_handle_t getInputDevice();
/* Returns the ID of the audio device actually used by the input to which this AudioRecord
* is attached.
* A value of AUDIO_PORT_HANDLE_NONE indicates the AudioRecord is not attached to any input.
*
* Parameters:
* none.
*/
audio_port_handle_t getRoutedDeviceId();
/* Add an AudioDeviceCallback. The caller will be notified when the audio device
* to which this AudioRecord is routed is updated.
* Replaces any previously installed callback.
* Parameters:
* callback: The callback interface
* Returns NO_ERROR if successful.
* INVALID_OPERATION if the same callback is already installed.
* NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable
* BAD_VALUE if the callback is NULL
*/
status_t addAudioDeviceCallback(
const sp<AudioSystem::AudioDeviceCallback>& callback);
/* remove an AudioDeviceCallback.
* Parameters:
* callback: The callback interface
* Returns NO_ERROR if successful.
* INVALID_OPERATION if the callback is not installed
* BAD_VALUE if the callback is NULL
*/
status_t removeAudioDeviceCallback(
const sp<AudioSystem::AudioDeviceCallback>& callback);
private:
/* If nonContig is non-NULL, it is an output parameter that will be set to the number of
* additional non-contiguous frames that are predicted to be available immediately,
* if the client were to release the first frames and then call obtainBuffer() again.
* This value is only a prediction, and needs to be confirmed.
* It will be set to zero for an error return.
* 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);
public:
/* Public API for TRANSFER_OBTAIN mode.
* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill.
*
* Buffer fields:
* frameCount currently ignored but recommend to set to actual number of frames consumed
* size actual number of bytes consumed, must be multiple of frameSize
* raw ignored
*/
void releaseBuffer(const 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().
* Default behavior is to only return when all data has been transferred. Set 'blocking' to
* false for the method to return immediately without waiting to try multiple times to read
* the full content of the buffer.
*/
ssize_t read(void* buffer, size_t size, bool blocking = true);
/* 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.
* FIXME The side-effect of resetting the counter may be incompatible with multi-client.
* Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
*/
uint32_t getInputFramesLost() const;
/* Get the flags */
audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; }
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
void wake(); // wake to handle changed notification conditions.
private:
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
bool mIgnoreNextPausedInt; // skip any internal pause and go immediately
// to processAudioBuffer() as state may have changed
// since pause time calculated.
};
// 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();
// caller must hold lock on mLock for all _l methods
status_t openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName);
// 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
// as specified in constructor or set()
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback
bool mRefreshRemaining; // processAudioBuffer() should refresh
// mRemainingFrames and mRetryOnPartialBuffer
// 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()
uint32_t mObservedSequence; // last observed value of mSequence
Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
Modulo<uint32_t> mNewPosition; // in frames
uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
status_t mStatus;
String16 mOpPackageName; // The package name used for app ops.
size_t mFrameCount; // corresponds to current IAudioRecord, value is
// reported back by AudioFlinger to the client
size_t mReqFrameCount; // frame count to request the first or next time
// a new IAudioRecord is needed, non-decreasing
int64_t mFramesRead; // total frames read. reset to zero after
// the start() following stop(). It is not
// changed after restoring the track.
int64_t mFramesReadServerOffset; // An offset to server frames read due to
// restoring AudioRecord, or stop/start.
// constant after constructor or set()
uint32_t mSampleRate;
audio_format_t mFormat;
uint32_t mChannelCount;
size_t mFrameSize; // app-level frame size == AudioFlinger frame size
uint32_t mLatency; // in ms
audio_channel_mask_t mChannelMask;
audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may
// be denied by client or server, such as
// AUDIO_INPUT_FLAG_FAST. mLock must be
// held to read or write those bits reliably.
audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const
audio_session_t mSessionId;
transfer_type mTransfer;
// Next 5 fields may be changed if IAudioRecord is re-created, but always != 0
// provided the initial set() was successful
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
sp<IMemory> mBufferMemory;
audio_io_handle_t mInput; // returned by AudioSystem::getInput()
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
private:
class DeathNotifier : public IBinder::DeathRecipient {
public:
DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { }
protected:
virtual void binderDied(const wp<IBinder>& who);
private:
const wp<AudioRecord> mAudioRecord;
};
sp<DeathNotifier> mDeathNotifier;
uint32_t mSequence; // incremented for each new IAudioRecord attempt
uid_t mClientUid;
pid_t mClientPid;
audio_attributes_t mAttributes;
// For Device Selection API
// a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing.
audio_port_handle_t mSelectedDeviceId;
sp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
audio_port_handle_t mPortId; // unique ID allocated by audio policy
};
}; // namespace android
#endif // ANDROID_AUDIORECORD_H