audiosystem分析之audiotrack源码
audiotrack.h分析 接下来会详细分析哈
/*
* Copyright (C) 2007 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_AUDIOTRACK_H
#define ANDROID_AUDIOTRACK_H
//定义audiotrack.h头文件
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
//将需要的头文件包含进来
namespace android { //android命名空间，同一个命名空间内，实现数据共享。
// ----------------------------------------------------------------------------
class audio_track_cblk_t;
// ----------------------------------------------------------------------------
class AudioTrack : virtual public RefBase //继承RefBase类 ，应用强弱指针。
{
public:
enum channel_index { //声道的枚举
MONO = 0,
LEFT = 0,
RIGHT = 1
};
/* Events used by AudioTrack callback function (audio_track_cblk_t).
*/
enum event_type { //流事件的枚举
EVENT_MORE_DATA = 0, // Request to write more data to PCM buffer.
EVENT_UNDERRUN = 1, // PCM buffer underrun occured.
EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from loop start if loop count was not 0.
EVENT_MARKER = 3, // Playback head is at the specified marker position (See setMarkerPosition()).
EVENT_NEW_POS = 4, // Playback head is at a new position (See setPositionUpdatePeriod()).
EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer.
};
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
* and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
class Buffer //Buffer类，实现音频buffer.
{
public:
enum {
MUTE = 0x00000001
};
uint32_t flags; // 0 or MUTE
audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
// accessed directly by WebKit ANP callback
int channelCount; // will be removed in the future, do not use
size_t frameCount; // number of sample frames corresponding to size;
// on input it is the number of frames desired,
// on output is the number of frames actually filled
size_t size; // input/output in byte units
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 available or various conditions occur.
* Parameters:
*
* event: type of event notified (see enum AudioTrack::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 AudioTrack::Buffer struct. The callback must not write
* more bytes than indicated by 'size' field and update 'size' if fewer bytes are
* written.
* - EVENT_UNDERRUN: unused.
* - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
* - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
* - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
* - EVENT_BUFFER_END: unused.
*/
typedef void (*callback_t)(int event, void* user, void *info);
/* Returns the minimum frame count required for the successful creation of
* an AudioTrack object.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
*/
static status_t getMinFrameCount(int* frameCount,
audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
uint32_t sampleRate = 0); //获得最小帧数
/* Constructs an uninitialized AudioTrack. No connection with
* AudioFlinger takes place.
*/
AudioTrack(); //audiotrack类的构造函数
/* Creates an audio 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:
*
* streamType: Select the type of audio stream this track is attached to
* (e.g. AUDIO_STREAM_MUSIC).
* 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: see audio_channels_t.
* frameCount: Minimum size of track PCM buffer in frames. This defines the
* latency of the track. The actual size selected by the AudioTrack could be
* larger if the requested size is not compatible with current audio HAL
* latency. Zero means to use a default value.
* flags: See comments on audio_output_flags_t in
* cbf: Callback function. If not null, this function is called periodically
* to request new PCM data.
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames have been consumed from track input buffer.
* sessionId: Specific session ID, or zero to use default.
* threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI.
* If not present in parameter list, then fixed at false.
*/
AudioTrack( audio_stream_type_t streamType,
uint32_t sampleRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
int sessionId = 0); //audiotrack类的构造函数
// DEPRECATED
explicit AudioTrack( int streamType,
uint32_t sampleRate = 0,
int format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
uint32_t flags = (uint32_t) AUDIO_OUTPUT_FLAG_NONE,
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
int sessionId = 0); //audiotrack类的构造函数
/* Creates an audio track and registers it with AudioFlinger. With this constructor,
* the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
* identified by the argument sharedBuffer. This prototype is for static buffer playback.
* PCM data must be present in memory before the AudioTrack is started.
* The write() and flush() methods are not supported in this case.
* It is recommended to pass a callback function to be notified of playback end by an
* EVENT_UNDERRUN event.
*/
AudioTrack( audio_stream_type_t streamType,
uint32_t sampleRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
const sp
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
int sessionId = 0); //audiotrack类的构造函数
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
*/
~AudioTrack();
//audiotrack类的析构函数
/* Initialize an uninitialized AudioTrack.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful initialization
* - INVALID_OPERATION: AudioTrack is already initialized
* - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* */
status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
uint32_t sampleRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
const sp
bool threadCanCallJava = false,
int sessionId = 0); //audio的设置
/* Result of constructing the AudioTrack. This must be checked
* before using any AudioTrack API (except for set()), because using
* an uninitialized AudioTrack 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 AudioTrack buffer size, AudioMixer (if any)
* and audio hardware driver.
*/
uint32_t latency() const;
/* getters, see constructors and set() */
audio_stream_type_t streamType() const;
audio_format_t format() const;
int channelCount() const;
uint32_t frameCount() const;
/* Return channelCount * (bit depth per channel / 8).
* channelCount is determined from channelMask, and bit depth comes from format.
*/
size_t frameSize() const;
sp
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
*/
void 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.
*/
void stop();
bool stopped() const;
/* Flush a stopped track. All pending buffers are discarded.
* This function has no effect if the track is not stopped.
*/
void flush();
/* Pause 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.
*/
void pause();
/* Mute or unmute this track.
* While muted, the callback, if set, is still called.
*/
void mute(bool);
bool muted() const;
/* Set volume for this track, mostly used for games' sound effects
* left and right volumes. Levels must be >= 0.0 and <= 1.0.
*/
status_t setVolume(float left, float right);
void getVolume(float* left, float* right) const;
/* Set the send level for this track. An auxiliary effect should be attached
* to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
*/
status_t setAuxEffectSendLevel(float level);
void getAuxEffectSendLevel(float* level) const;
/* Set sample rate for this track, mostly used for games' sound effects
*/
status_t setSampleRate(int sampleRate);
uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
*
* Parameters:
*
* loopStart: loop start expressed as the number of PCM frames played since AudioTrack start.
* loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start.
* loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
* pending or active loop. loopCount = -1 means infinite looping.
*
* For proper operation the following condition must be respected:
* (loopEnd-loopStart) <= framecount()
*/
status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
/* Sets marker position. When playback 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 AudioTrack 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 AudioTrack 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 played,
* a callback with event type EVENT_NEW_POS is called.
* Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
* callback.
* If the AudioTrack 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 AudioTrack has no callback installed.
*/
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
/* Sets playback head position within AudioTrack buffer. The new position is specified
* in number of frames.
* This method must be called with the AudioTrack in paused or stopped state.
* Note that the actual position set is
* Therefore using this method makes sense only when playing a "static" audio buffer
* as opposed to streaming.
* The getPosition() method on the other hand returns the total number of frames played since
* playback start.
*
* Parameters:
*
* position: New playback head position within AudioTrack buffer.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack is not stopped.
* - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
*/
status_t setPosition(uint32_t position);
status_t getPosition(uint32_t *position);
/* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
* rewriting the buffer before restarting playback after a stop.
* This method must be called with the AudioTrack in paused or stopped state.
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack is not stopped.
*/
status_t reload();
/* Returns a handle on the audio output used by this AudioTrack.
*
* Parameters:
* none.
*
* Returned value:
* handle on audio hardware output
*/
audio_io_handle_t getOutput();
/* Returns the unique session ID associated with this track.
*
* Parameters:
* none.
*
* Returned value:
* AudioTrack session ID.
*/
int getSessionId() const;
/* Attach track auxiliary output to specified effect. Use effectId = 0
* to detach track from effect.
*
* Parameters:
*
* effectId: effectId obtained from AudioEffect::id().
*
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the effect is not an auxiliary effect.
* - BAD_VALUE: The specified effect ID is invalid
*/
status_t attachAuxEffect(int effectId);
/* Obtains a buffer of "frameCount" frames. The buffer must be
* filled 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 (buffercount())
* 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 a filled buffer of "frameCount" frames for AudioFlinger to process. */
void releaseBuffer(Buffer* audioBuffer);
/* As a convenience we provide a write() interface to the audio buffer.
* This is implemented on top of obtainBuffer/releaseBuffer. For best
* performance use callbacks. Returns actual number of bytes written >= 0,
* or one of the following negative status codes:
* INVALID_OPERATION AudioTrack is configured for shared buffer mode
* BAD_VALUE size is invalid
* STOPPED AudioTrack was stopped during the write
* NO_MORE_BUFFERS when obtainBuffer() returns same
* or any other error code returned by IAudioTrack::start() or restoreTrack_l().
*/
ssize_t write(const void* buffer, size_t size);
/*
* Dumps the state of an audio track.
*/
status_t dump(int fd, const Vector
protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
AudioTrack& operator = (const AudioTrack& other);
/* a small internal class to handle the callback */
class AudioTrackThread : public Thread
{
public:
AudioTrackThread(AudioTrack& 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 AudioTrack;
virtual bool threadLoop();
virtual status_t readyToRun();
virtual void onFirstRef();
AudioTrack& mReceiver;
~AudioTrackThread();
Mutex mMyLock; // Thread::mLock is private
Condition mMyCond; // Thread::mThreadExitedCondition is private
bool mPaused; // whether thread is currently paused
};
// body of AudioTrackThread::threadLoop()
bool processAudioBuffer(const sp
status_t createTrack_l(audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
audio_output_flags_t flags,
const sp
audio_io_handle_t output);
void flush_l();
status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
audio_io_handle_t getOutput_l();
status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
bool stopped_l() const { return !mActive; }
sp
sp
sp
float mVolume[2];
float mSendLevel;
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
audio_format_t mFormat;
audio_stream_type_t mStreamType;
uint8_t mChannelCount;
uint8_t mMuted;
uint8_t mReserved;
uint32_t mChannelMask;
status_t mStatus;
uint32_t mLatency;
bool mActive; // protected by mLock
callback_t mCbf; // callback handler for events, or NULL
void* mUserData;
uint32_t mNotificationFramesReq; // requested number of frames between each notification callback
uint32_t mNotificationFramesAct; // actual number of frames between each notification callback
sp
int mLoopCount;
uint32_t mRemainingFrames;
uint32_t mMarkerPosition;
bool mMarkerReached;
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
bool mFlushed; // FIXME will be made obsolete by making flush() synchronous
audio_output_flags_t mFlags;
int mSessionId;
int mAuxEffectId;
mutable Mutex mLock;
status_t mRestoreStatus;
bool mIsTimed;
int mPreviousPriority; // before start()
SchedPolicy mPreviousSchedulingGroup;
};
class TimedAudioTrack : public AudioTrack
{
public:
TimedAudioTrack();
/* allocate a shared memory buffer that can be passed to queueTimedBuffer */
status_t allocateTimedBuffer(size_t size, sp
/* queue a buffer obtained via allocateTimedBuffer for playback at the
given timestamp. PTS units a microseconds on the media time timeline.
The media time transform (set with setMediaTimeTransform) set by the
audio producer will handle converting from media time to local time
(perhaps going through the common time timeline in the case of
synchronized multiroom audio case) */
status_t queueTimedBuffer(const sp
/* define a transform between media time and either common time or
local time */
enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
status_t setMediaTimeTransform(const LinearTransform& xform,
TargetTimeline target);
};
}; // namespace android
#endif // ANDROID_AUDIOTRACK_H
以上为audioTrack.h文件的分析，audiotrack.cpp是audiotrack.h中声明的函数的实现。实现方法如下：
audiotrack.cpp分析
/*
**** Copyright 2007, 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.
*/
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioTrack"
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
namespace android { //实现audiotrack类，该类主要实现声道、缓存、流类型等信息的定义，及音频流的播放等。
// ---------------------------------------------------------------------------
// static
status_t AudioTrack::getMinFrameCount(
int* frameCount,
audio_stream_type_t streamType,
uint32_t sampleRate)
{
// FIXME merge with similar code in createTrack_l(), except we're missing
// some information here that is available in createTrack_l():
// audio_io_handle_t output
// audio_format_t format
// audio_channel_mask_t channelMask
// audio_output_flags_t flags
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
if (minBufCount < 2) minBufCount = 2;
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
*frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
return NO_ERROR;
}
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
: mStatus(NO_INIT),
mIsTimed(false),androidsource
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT)
{
}
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
int channelMask,
int frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
}
// DEPRECATED
AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
int channelMask,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
}
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
int channelMask,
const sp
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
sharedBuffer, false /*threadCanCallJava*/, sessionId);
}
AudioTrack::~AudioTrack()
{
ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
if (mStatus == NO_ERROR) {
// Make sure that callback function exits in the case where
// it is looping on buffer full condition in obtainBuffer().
// Otherwise the callback thread will never exit.
stop();
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
mAudioTrack.clear();
IPCThreadState::self()->flushCommands();
AudioSystem::releaseAudioSessionId(mSessionId);
}
}
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
int channelMask,
int frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
const sp
bool threadCanCallJava,
int sessionId)
{
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
if (mAudioTrack != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (sampleRate == 0) {
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channelMask == 0) {
channelMask = AUDIO_CHANNEL_OUT_STEREO;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format");
return BAD_VALUE;
}
// AudioFlinger does not currently support 8-bit data in shared memory
if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
ALOGE("8-bit data in shared memory is not supported");
return BAD_VALUE;
}
// force direct flag if format is not linear PCM
if (!audio_is_linear_pcm(format)) {
flags = (audio_output_flags_t)
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// only allow deep buffering for music stream type
if (streamType != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask");
return BAD_VALUE;
}
uint32_t channelCount = popcount(channelMask);
audio_io_handle_t output = AudioSystem::getOutput(
streamType,
sampleRate, format, channelMask,
flags);
if (output == 0) {
ALOGE("Could not get audio output for stream type %d", streamType);
return BAD_VALUE;
}
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0.0f;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
mAuxEffectId = 0;
mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
}
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
format,
(uint32_t)channelMask,
frameCount,
flags,
sharedBuffer,
output);
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit();
mAudioTrackThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mStreamType = streamType;
mFormat = format;
mChannelMask = (uint32_t)channelMask;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = false;
mUserData = user;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mFlushed = false;
AudioSystem::acquireAudioSessionId(mSessionId);
mRestoreStatus = NO_ERROR;
return NO_ERROR;
}
status_t AudioTrack::initCheck() const
{
return mStatus;
}
// -------------------------------------------------------------------------
uint32_t AudioTrack::latency() const
{
return mLatency;
}
audio_stream_type_t AudioTrack::streamType() const
{
return mStreamType;
}
audio_format_t AudioTrack::format() const
{
return mFormat;
}
int AudioTrack::channelCount() const
{
return mChannelCount;
}
uint32_t AudioTrack::frameCount() const
{
return mCblk->frameCount;
}
size_t AudioTrack::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
} else {
return sizeof(uint8_t);
}
}
sp
{
return mSharedBuffer;
}
// -------------------------------------------------------------------------
void AudioTrack::start()
{
sp
status_t status = NO_ERROR;
ALOGV("start %p", this);
AutoMutex lock(mLock);
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
sp
sp
audio_track_cblk_t* cblk = mCblk;
if (!mActive) {
mFlushed = false;
mActive = true;
mNewPosition = cblk->server + mUpdatePeriod;
cblk->lock.lock();
cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
cblk->waitTimeMs = 0;
android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
if (t != 0) {
t->resume();
} else {
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
get_sched_policy(0, &mPreviousSchedulingGroup);
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
ALOGV("start %p before lock cblk %p", this, mCblk);
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
ALOGV("mAudioTrack->start()");
status = mAudioTrack->start();
cblk->lock.lock();
if (status == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
}
}
if (cblk->flags & CBLK_INVALID_MSK) {
status = restoreTrack_l(cblk, true);
}
cblk->lock.unlock();
if (status != NO_ERROR) {
ALOGV("start() failed");
mActive = false;
if (t != 0) {
t->pause();
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
}
}
void AudioTrack::stop()
{
sp
ALOGV("stop %p", this);
AutoMutex lock(mLock);
if (mActive) {
mActive = false;
mCblk->cv.signal();
mAudioTrack->stop();
// Cancel loops (If we are in the middle of a loop, playback
// would not stop until loopCount reaches 0).
setLoop_l(0, 0, 0);
// the playback head position will reset to 0, so if a marker is set, we need
// to activate it again
mMarkerReached = false;
// Force flush if a shared buffer is used otherwise audioflinger
// will not stop before end of buffer is reached.
if (mSharedBuffer != 0) {
flush_l();
}
if (t != 0) {
t->pause();
} else {
setpriority(PRIO_PROCESS, 0, mPreviousPriority);
set_sched_policy(0, mPreviousSchedulingGroup);
}
}
}
bool AudioTrack::stopped() const
{
AutoMutex lock(mLock);
return stopped_l();
}
void AudioTrack::flush()
{
AutoMutex lock(mLock);
flush_l();
}
// must be called with mLock held
void AudioTrack::flush_l()
{
ALOGV("flush");
// clear playback marker and periodic update counter
mMarkerPosition = 0;
mMarkerReached = false;
mUpdatePeriod = 0;
if (!mActive) {
mFlushed = true;
mAudioTrack->flush();
// Release AudioTrack callback thread in case it was waiting for new buffers
// in AudioTrack::obtainBuffer()
mCblk->cv.signal();
}
}
void AudioTrack::pause()
{
ALOGV("pause");
AutoMutex lock(mLock);
if (mActive) {
mActive = false;
mCblk->cv.signal();
mAudioTrack->pause();
}
}
void AudioTrack::mute(bool e)
{
mAudioTrack->mute(e);
mMuted = e;
}
bool AudioTrack::muted() const
{
return mMuted;
}
status_t AudioTrack::setVolume(float left, float right)
{
if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
return NO_ERROR;
}
void AudioTrack::getVolume(float* left, float* right) const
{
if (left != NULL) {
*left = mVolume[LEFT];
}
if (right != NULL) {
*right = mVolume[RIGHT];
}
}
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
if (level < 0.0f || level > 1.0f) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mSendLevel = level;
mCblk->setSendLevel(level);
return NO_ERROR;
}
void AudioTrack::getAuxEffectSendLevel(float* level) const
{
if (level != NULL) {
*level = mSendLevel;
}
}
status_t AudioTrack::setSampleRate(int rate)
{
int afSamplingRate;
if (mIsTimed) {
return INVALID_OPERATION;
}
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
AutoMutex lock(mLock);
mCblk->sampleRate = rate;
return NO_ERROR;
}
uint32_t AudioTrack::getSampleRate() const
{
if (mIsTimed) {
return INVALID_OPERATION;
}
AutoMutex lock(mLock);
return mCblk->sampleRate;
}
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
AutoMutex lock(mLock);
return setLoop_l(loopStart, loopEnd, loopCount);
}
// must be called with mLock held
status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
audio_track_cblk_t* cblk = mCblk;
Mutex::Autolock _l(cblk->lock);
if (loopCount == 0) {
cblk->loopStart = UINT_MAX;
cblk->loopEnd = UINT_MAX;
cblk->loopCount = 0;
mLoopCount = 0;
return NO_ERROR;
}
if (mIsTimed) {
return INVALID_OPERATION;
}
if (loopStart >= loopEnd ||
loopEnd - loopStart > cblk->frameCount ||
cblk->server > loopStart) {
ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
return BAD_VALUE;
}
if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
loopStart, loopEnd, cblk->frameCount);
return BAD_VALUE;
}
cblk->loopStart = loopStart;
cblk->loopEnd = loopEnd;
cblk->loopCount = loopCount;
mLoopCount = loopCount;
return NO_ERROR;
}
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
return NO_ERROR;
}
status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
return NO_ERROR;
}
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
mNewPosition = curPosition + updatePeriod;
mUpdatePeriod = updatePeriod;
return NO_ERROR;
}
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
}
status_t AudioTrack::setPosition(uint32_t position)
{
if (mIsTimed) return INVALID_OPERATION;
AutoMutex lock(mLock);
if (!stopped_l()) return INVALID_OPERATION;
Mutex::Autolock _l(mCblk->lock);
if (position > mCblk->user) return BAD_VALUE;
mCblk->server = position;
android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
return NO_ERROR;
}
status_t AudioTrack::getPosition(uint32_t *position)
{
if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mFlushed ? 0 : mCblk->server;
return NO_ERROR;
}
status_t AudioTrack::reload()
{
AutoMutex lock(mLock);
if (!stopped_l()) return INVALID_OPERATION;
flush_l();
mCblk->stepUser(mCblk->frameCount);
return NO_ERROR;
}
audio_io_handle_t AudioTrack::getOutput()
{
AutoMutex lock(mLock);
return getOutput_l();
}
// must be called with mLock held
audio_io_handle_t AudioTrack::getOutput_l()
{
return AudioSystem::getOutput(mStreamType,
mCblk->sampleRate, mFormat, mChannelMask, mFlags);
}
int AudioTrack::getSessionId() const
{
return mSessionId;
}
status_t AudioTrack::attachAuxEffect(int effectId)
{
ALOGV("attachAuxEffect(%d)", effectId);
status_t status = mAudioTrack->attachAuxEffect(effectId);
if (status == NO_ERROR) {
mAuxEffectId = effectId;
}
return status;
}
// -------------------------------------------------------------------------
// must be called with mLock held
status_t AudioTrack::createTrack_l(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
audio_output_flags_t flags,
const sp
audio_io_handle_t output)
{
status_t status;
const sp
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
return NO_INIT;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
// either of these use cases:
// use case 1: shared buffer
(sharedBuffer != 0) ||
// use case 2: callback handler
(mCbf != NULL))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
// once denied, do not request again if IAudioTrack is re-created
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
}
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
mNotificationFramesAct = mNotificationFramesReq;
if (!audio_is_linear_pcm(format)) {
if (sharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
int afFrameCount;
if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
return NO_INIT;
}
frameCount = afFrameCount;
}
} else if (sharedBuffer != 0) {
// Ensure that buffer alignment matches channelCount
int channelCount = popcount(channelMask);
// 8-bit data in shared memory is not currently supported by AudioFlinger
size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
if (channelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channelCount %d",
sharedBuffer->pointer(), channelCount);
return BAD_VALUE;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
} else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
int afSampleRate;
if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
return NO_INIT;
}
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
if (minBufCount < 2) minBufCount = 2;
int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
", afLatency=%d",
minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
if (frameCount == 0) {
frameCount = minFrameCount;
}
if (mNotificationFramesAct == 0) {
mNotificationFramesAct = frameCount/2;
}
// Make sure that application is notified with sufficient margin
// before underrun
if (mNotificationFramesAct > (uint32_t)frameCount/2) {
mNotificationFramesAct = frameCount/2;
}
if (frameCount < minFrameCount) {
// not ALOGW because it happens all the time when playing key clicks over A2DP
ALOGV("Minimum buffer size corrected from %d to %d",
frameCount, minFrameCount);
frameCount = minFrameCount;
}
} else {
// For fast tracks, the frame count calculations and checks are done by server
}
IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
if (mIsTimed) {
trackFlags |= IAudioFlinger::TRACK_TIMED;
}
pid_t tid = -1;
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
sp
streamType,
sampleRate,
format,
channelMask,
frameCount,
trackFlags,
sharedBuffer,
output,
tid,
&mSessionId,
&status);
if (track == 0) {
ALOGE("AudioFlinger could not create track, status: %d", status);
return status;
}
sp
if (cblk == 0) {
ALOGE("Could not get control block");
return NO_INIT;
}
mAudioTrack = track;
mCblkMemory = cblk;
mCblk = static_cast
// old has the previous value of mCblk->flags before the "or" operation
int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
if (old & CBLK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount);
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount);
// once denied, do not request again if IAudioTrack is re-created
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
}
if (sharedBuffer == 0) {
mNotificationFramesAct = mCblk->frameCount/2;
}
}
if (sharedBuffer == 0) {
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
} else {
mCblk->buffers = sharedBuffer->pointer();
// Force buffer full condition as data is already present in shared memory
mCblk->stepUser(mCblk->frameCount);
}
mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
mCblk->setSendLevel(mSendLevel);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
mRemainingFrames = mNotificationFramesAct;
// FIXME don't believe this lie
mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (mCblk->frameCount > mFrameCount) {
mFrameCount = mCblk->frameCount;
}
return NO_ERROR;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
AutoMutex lock(mLock);
bool active;
status_t result = NO_ERROR;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
audioBuffer->frameCount = 0;
audioBuffer->size = 0;
uint32_t framesAvail = cblk->framesAvailable();
cblk->lock.lock();
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
cblk->lock.unlock();
if (framesAvail == 0) {
cblk->lock.lock();
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
if (!(cblk->flags & CBLK_INVALID_MSK)) {
mLock.unlock();
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
cblk->lock.unlock();
mLock.lock();
if (!mActive) {
return status_t(STOPPED);
}
cblk->lock.lock();
}
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
// timing out when a loop has been set and we have already written upto loop end
// is a normal condition: no need to wake AudioFlinger up.
if (cblk->user < cblk->loopEnd) {
ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p name=%#x"
"user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server);
//unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
cblk->lock.unlock();
result = mAudioTrack->start();
cblk->lock.lock();
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
create_new_track:
result = restoreTrack_l(cblk, false);
}
if (result != NO_ERROR) {
ALOGW("obtainBuffer create Track error %d", result);
cblk->lock.unlock();
return result;
}
}
cblk->waitTimeMs = 0;
}
if (--waitCount == 0) {
cblk->lock.unlock();
return TIMED_OUT;
}
}
// read the server count again
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
cblk->lock.unlock();
}
cblk->waitTimeMs = 0;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (framesReq > bufferEnd - u) {
framesReq = bufferEnd - u;
}
audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
audioBuffer->channelCount = mChannelCount;
audioBuffer->frameCount = framesReq;
audioBuffer->size = framesReq * cblk->frameSize;
if (audio_is_linear_pcm(mFormat)) {
audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
} else {
audioBuffer->format = mFormat;
}
audioBuffer->raw = (int8_t *)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
AutoMutex lock(mLock);
mCblk->stepUser(audioBuffer->frameCount);
if (audioBuffer->frameCount > 0) {
// restart track if it was disabled by audioflinger due to previous underrun
if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName);
mAudioTrack->start();
}
}
}
// -------------------------------------------------------------------------
ssize_t AudioTrack::write(const void* buffer, size_t userSize)
{
if (mSharedBuffer != 0) return INVALID_OPERATION;
if (mIsTimed) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
}
ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
if (userSize == 0) {
return 0;
}
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
mLock.lock();
sp
sp
mLock.unlock();
ssize_t written = 0;
const int8_t *src = (const int8_t *)buffer;
Buffer audioBuffer;
size_t frameSz = frameSize();
do {
audioBuffer.frameCount = userSize/frameSz;
status_t err = obtainBuffer(&audioBuffer, -1);
if (err < 0) {
// out of buffers, return #bytes written
if (err == status_t(NO_MORE_BUFFERS))
break;
return ssize_t(err);
}
size_t toWrite;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
} else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
src += toWrite;
}
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
} while (userSize >= frameSz);
return written;
}
// -------------------------------------------------------------------------
TimedAudioTrack::TimedAudioTrack() {
mIsTimed = true;
}
status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp
{
status_t result = UNKNOWN_ERROR;
// If the track is not invalid already, try to allocate a buffer. alloc
// fails indicating that the server is dead, flag the track as invalid so
// we can attempt to restore in in just a bit.
if (!(mCblk->flags & CBLK_INVALID_MSK)) {
result = mAudioTrack->allocateTimedBuffer(size, buffer);
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
}
}
// If the track is invalid at this point, attempt to restore it. and try the
// allocation one more time.
if (mCblk->flags & CBLK_INVALID_MSK) {
mCblk->lock.lock();
result = restoreTrack_l(mCblk, false);
mCblk->lock.unlock();
if (result == OK)
result = mAudioTrack->allocateTimedBuffer(size, buffer);
}
return result;
}
status_t TimedAudioTrack::queueTimedBuffer(const sp
int64_t pts)
{
status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
{
AutoMutex lock(mLock);
// restart track if it was disabled by audioflinger due to previous underrun
if (buffer->size() != 0 && status == NO_ERROR &&
mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
mAudioTrack->start();
}
}
return status;
}
status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
TargetTimeline target)
{
return mAudioTrack->setMediaTimeTransform(xform, target);
}
// -------------------------------------------------------------------------
bool AudioTrack::processAudioBuffer(const sp
{
Buffer audioBuffer;
uint32_t frames;
size_t writtenSize;
mLock.lock();
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
sp
sp
audio_track_cblk_t* cblk = mCblk;
bool active = mActive;
mLock.unlock();
// Manage underrun callback
if (active && (cblk->framesAvailable() == cblk->frameCount)) {
ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
if (cblk->server == cblk->frameCount) {
mCbf(EVENT_BUFFER_END, mUserData, 0);
}
if (mSharedBuffer != 0) return false;
}
}
// Manage loop end callback
while (mLoopCount > cblk->loopCount) {
int loopCount = -1;
mLoopCount--;
if (mLoopCount >= 0) loopCount = mLoopCount;
mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
}
// Manage marker callback
if (!mMarkerReached && (mMarkerPosition > 0)) {
if (cblk->server >= mMarkerPosition) {
mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
mMarkerReached = true;
}
}
// Manage new position callback
if (mUpdatePeriod > 0) {
while (cblk->server >= mNewPosition) {
mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
mNewPosition += mUpdatePeriod;
}
}
// If Shared buffer is used, no data is requested from client.
if (mSharedBuffer != 0) {
frames = 0;
} else {
frames = mRemainingFrames;
}
// See description of waitCount parameter at declaration of obtainBuffer().
// The logic below prevents us from being stuck below at obtainBuffer()
// not being able to handle timed events (position, markers, loops).
int32_t waitCount = -1;
if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
waitCount = 1;
}
do {
audioBuffer.frameCount = frames;
status_t err = obtainBuffer(&audioBuffer, waitCount);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
return false;
}
break;
}
if (err == status_t(STOPPED)) return false;
// Divide buffer size by 2 to take into account the expansion
// due to 8 to 16 bit conversion: the callback must fill only half
// of the destination buffer
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
audioBuffer.size >>= 1;
}
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
writtenSize = audioBuffer.size;
// Sanity check on returned size
if (ssize_t(writtenSize) <= 0) {
// The callback is done filling buffers
// Keep this thread going to handle timed events and
// still try to get more data in intervals of WAIT_PERIOD_MS
// but don't just loop and block the CPU, so wait
usleep(WAIT_PERIOD_MS*1000);
break;
}
if (writtenSize > reqSize) writtenSize = reqSize;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// 8 to 16 bit conversion, note that source and destination are the same address
memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
writtenSize <<= 1;
}
audioBuffer.size = writtenSize;
// NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit.
audioBuffer.frameCount = writtenSize/mCblk->frameSize;
frames -= audioBuffer.frameCount;
releaseBuffer(&audioBuffer);
}
while (frames);
if (frames == 0) {
mRemainingFrames = mNotificationFramesAct;
} else {
mRemainingFrames = frames;
}
return true;
}
// must be called with mLock and cblk.lock held. Callers must also hold strong references on
// the IAudioTrack and IMemory in case they are recreated here.
// If the IAudioTrack is successfully restored, the cblk pointer is updated
status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
{
status_t result;
if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
fromStart ? "start()" : "obtainBuffer()", gettid());
// signal old cblk condition so that other threads waiting for available buffers stop
// waiting now
cblk->cv.broadcast();
cblk->lock.unlock();
// refresh the audio configuration cache in this process to make sure we get new
// output parameters in getOutput_l() and createTrack_l()
AudioSystem::clearAudioConfigCache();
// if the new IAudioTrack is created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory
result = createTrack_l(mStreamType,
cblk->sampleRate,
mFormat,
mChannelMask,
mFrameCount,
mFlags,
mSharedBuffer,
getOutput_l());
if (result == NO_ERROR) {
uint32_t user = cblk->user;
uint32_t server = cblk->server;
// restore write index and set other indexes to reflect empty buffer status
mCblk->user = user;
mCblk->server = user;
mCblk->userBase = user;
mCblk->serverBase = user;
// restore loop: this is not guaranteed to succeed if new frame count is not
// compatible with loop length
setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
if (!fromStart) {
mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
// Make sure that a client relying on callback events indicating underrun or
// the actual amount of audio frames played (e.g SoundPool) receives them.
if (mSharedBuffer == 0) {
uint32_t frames = 0;
if (user > server) {
frames = ((user - server) > mCblk->frameCount) ?
mCblk->frameCount : (user - server);
memset(mCblk->buffers, 0, frames * mCblk->frameSize);
}
// restart playback even if buffer is not completely filled.
android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
// stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
// the client
mCblk->stepUser(frames);
}
}
if (mSharedBuffer != 0) {
mCblk->stepUser(mCblk->frameCount);
}
if (mActive) {
result = mAudioTrack->start();
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
}
if (fromStart && result == NO_ERROR) {
mNewPosition = mCblk->server + mUpdatePeriod;
}
}
if (result != NO_ERROR) {
android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
}
mRestoreStatus = result;
// signal old cblk condition for other threads waiting for restore completion
android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
cblk->cv.broadcast();
} else {
if (!(cblk->flags & CBLK_RESTORED_MSK)) {
ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid());
mLock.unlock();
result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
if (result == NO_ERROR) {
result = mRestoreStatus;
}
cblk->lock.unlock();
mLock.lock();
} else {
ALOGW("dead IAudioTrack, already restored TID %d", gettid());
result = mRestoreStatus;
cblk->lock.unlock();
}
}
ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
if (result == NO_ERROR) {
// from now on we switch to the newly created cblk
cblk = mCblk;
}
cblk->lock.lock();
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
return result;
}
status_t AudioTrack::dump(int fd, const Vector
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append(" AudioTrack::dump\n");
snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
result.append(buffer);
snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
result.append(buffer);
snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
// =========================================================================
AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
: Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
{
}
AudioTrack::AudioTrackThread::~AudioTrackThread()
{
}
bool AudioTrack::AudioTrackThread::threadLoop()
{
{
AutoMutex _l(mMyLock);
if (mPaused) {
mMyCond.wait(mMyLock);
// caller will check for exitPending()
return true;
}
}
if (!mReceiver.processAudioBuffer(this)) {
pause();
}
return true;
}
status_t AudioTrack::AudioTrackThread::readyToRun()
{
return NO_ERROR;
}
void AudioTrack::AudioTrackThread::onFirstRef()
{
}
void AudioTrack::AudioTrackThread::requestExit()
{
// must be in this order to avoid a race condition
Thread::requestExit();
resume();
}
void AudioTrack::AudioTrackThread::pause()
{
AutoMutex _l(mMyLock);
mPaused = true;
}
void AudioTrack::AudioTrackThread::resume()
{
AutoMutex _l(mMyLock);
if (mPaused) {
mPaused = false;
mMyCond.signal();
}
}
// =========================================================================
audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
userBase(0), serverBase(0), buffers(NULL), frameCount(0),
loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
mSendLevel(0), flags(0)
{
}
uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
ALOGV("stepuser %08x %08x %d", user, server, frameCount);
uint32_t u = user;
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
if (flags & CBLK_DIRECTION_MSK) {
// If stepServer() has been called once, switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
}
} else if (u > server) {
ALOGW("stepUser occurred after track reset");
u = server;
}
uint32_t fc = this->frameCount;
if (u >= fc) {
// common case, user didn't just wrap
if (u - fc >= userBase ) {
userBase += fc;
}
} else if (u >= userBase + fc) {
// user just wrapped
userBase += fc;
}
user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
if (flags & CBLK_UNDERRUN_MSK) {
android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
}
return u;
}
bool audio_track_cblk_t::stepServer(uint32_t frameCount)
{
ALOGV("stepserver %08x %08x %d", user, server, frameCount);
if (!tryLock()) {
ALOGW("stepServer() could not lock cblk");
return false;
}
uint32_t s = server;
bool flushed = (s == user);
s += frameCount;
if (flags & CBLK_DIRECTION_MSK) {
// Mark that we have read the first buffer so that next time stepUser() is called
// we switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
}
// It is possible that we receive a flush()
// while the mixer is processing a block: in this case,
// stepServer() is called After the flush() has reset u & s and
// we have s > u
if (flushed) {
ALOGW("stepServer occurred after track reset");
s = user;
}
}
if (s >= loopEnd) {
ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
s = loopStart;
if (--loopCount == 0) {
loopEnd = UINT_MAX;
loopStart = UINT_MAX;
}
}
uint32_t fc = this->frameCount;
if (s >= fc) {
// common case, server didn't just wrap
if (s - fc >= serverBase ) {
serverBase += fc;
}
} else if (s >= serverBase + fc) {
// server just wrapped
serverBase += fc;
}
server = s;
if (!(flags & CBLK_INVALID_MSK)) {
cv.signal();
}
lock.unlock();
return true;
}
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
return (int8_t *)buffers + (offset - userBase) * frameSize;
}
uint32_t audio_track_cblk_t::framesAvailable()
{
Mutex::Autolock _l(lock);
return framesAvailable_l();
}
uint32_t audio_track_cblk_t::framesAvailable_l()
{
uint32_t u = user;
uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
return limit + frameCount - u;
} else {
return frameCount + u - s;
}
}
uint32_t audio_track_cblk_t::framesReady()
{
uint32_t u = user;
uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
return u - s;
} else {
// do not block on mutex shared with client on AudioFlinger side
if (!tryLock()) {
ALOGW("framesReady() could not lock cblk");
return 0;
}
uint32_t frames = UINT_MAX;
if (loopCount >= 0) {
frames = (loopEnd - loopStart)*loopCount + u - s;
}
lock.unlock();
return frames;
}
} else {
return s - u;
}
}
bool audio_track_cblk_t::tryLock()
{
// the code below simulates lock-with-timeout
// we MUST do this to protect the AudioFlinger server
// as this lock is shared with the client.
status_t err;
err = lock.tryLock();
if (err == -EBUSY) { // just wait a bit
usleep(1000);
err = lock.tryLock();
}
if (err != NO_ERROR) {
// probably, the client just died.
return false;
}
return true;
}
// -------------------------------------------------------------------------
}; // namespace android