在上一篇介绍了关于AudioPolicyService与AudioFlinger服务的启动及初始化。这里探索一下AudioTrack与AudioFlinger做了那些事情。
MediaPlayer会在framework层创建相应的音频解码器。而AudioTrack仅仅能播放已经解码的PCM数据流，MediaPlayer在framework层还会创建AudioTrack，把解码后的PCM数流传递给AudioTrack。AudioTrack再传递给AudioFlinger进行混音，然后才传递给硬件播放,所以是MediaPlayer包括了AudioTrack。
例：
AudioTrack audio = new AudioTrack(
AudioManager.STREAM_MUSIC, // 指定流的类型
32000, // 设置音频数据的採样率 32k，假设是44.1k就是44100
AudioFormat.CHANNEL_OUT_STEREO, // 设置输出声道为双声道立体声，而CHANNEL_OUT_MONO类型是单声道
AudioFormat.ENCODING_PCM_16BIT, // 设置音频数据块是8位还是16位。这里设置为16位。
好像如今绝大多数的音频都是16位的了
AudioTrack.MODE_STREAM // 设置模式类型，在这里设置为流类型，第二种MODE_STATIC貌似没有什么效果
);
audio.play(); // 启动音频设备。以下就能够真正開始音频数据的播放了
// 打开mp3文件，读取数据，解码等操作省略 ...
byte[] buffer = new buffer[4096];
int count;
while(true)
{
// 最关键的是将解码后的数据，从缓冲区写入到AudioTrack对象中
audio.write(buffer, 0, 4096);
if(文件结束) break;
}
//关闭并释放资源
audio.stop();
audio.release();每个音频流都会创建AudioTrack类的一个实例，每个AudioTrack会在创建时注冊到 AudioFlinger中。由AudioFlinger把全部的AudioTrack进行混合（Mixer）。然后输送到 AudioHardware中进行播放。 (网上资源)
下面我们分析一下源码：
AudioTtack.java
/**
* streamType:音频流类型
* sampleRateInHz:採样率
* channelConfig:音频声道
* audioFormat:音频格式
* bufferSizeInBytes缓冲区大小:
* mode:音频数据载入模式
* sessionId:会话id
*/
private AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int mode, int sessionId, boolean offload, int encapsulationMode,
@Nullable TunerConfiguration tunerConfiguration)
throws IllegalArgumentException {
super(attributes, AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
// mState already == STATE_UNINITIALIZED
mConfiguredAudioAttributes = attributes; // object copy not needed, immutable.
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat");
}
// Check if we should enable deep buffer mode(检查我们是否应该启用深度缓冲模式)
if (shouldEnablePowerSaving(mAttributes, format, bufferSizeInBytes, mode)) {
mAttributes = new AudioAttributes.Builder(mAttributes)
.replaceFlags((mAttributes.getAllFlags()
| AudioAttributes.FLAG_DEEP_BUFFER)
& ~AudioAttributes.FLAG_LOW_LATENCY)
.build();
}
// remember which looper is associated with the AudioTrack instantiation
//记住哪个循环器与 AudioTrack 实例化相关联
Looper looper;
if ((looper = Looper.myLooper()) == null) {
looper = Looper.getMainLooper();
}
int rate = format.getSampleRate();
if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
rate = 0;
}
int channelIndexMask = 0;
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {
channelIndexMask = format.getChannelIndexMask();
}
int channelMask = 0;
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {
channelMask = format.getChannelMask();
} else if (channelIndexMask == 0) { // if no masks at all, use stereo
channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT
| AudioFormat.CHANNEL_OUT_FRONT_RIGHT;
}
int encoding = AudioFormat.ENCODING_DEFAULT;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {
encoding = format.getEncoding();
}
/**
* 參数检查
* 1.检查streamType是否为：STREAM_ALARM、STREAM_MUSIC、STREAM_RING、STREAM_SYSTEM、STREAM_VOICE_CALL、
* STREAM_NOTIFICATION、STREAM_BLUETOOTH_SCO、STREAM_BLUETOOTH_SCO，并赋值给mStreamType
* 2.检查sampleRateInHz是否在4000到48000之间。并赋值给mSampleRate
* 3.设置mChannels：
* CHANNEL_OUT_DEFAULT、CHANNEL_OUT_MONO、CHANNEL_CONFIGURATION_MONO ---> CHANNEL_OUT_MONO
* CHANNEL_OUT_STEREO、CHANNEL_CONFIGURATION_STEREO ---> CHANNEL_OUT_STEREO
* 4.设置mAudioFormat：
* ENCODING_PCM_16BIT、ENCODING_DEFAULT ---> ENCODING_PCM_16BIT
* ENCODING_PCM_8BIT ---> ENCODING_PCM_8BIT
* 5.设置mDataLoadMode：
* MODE_STREAM
* MODE_STATIC
*/
audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode);
mOffloaded = offload;
mStreamType = AudioSystem.STREAM_DEFAULT;
/**
* buffer大小检查,计算每帧字节大小，假设是ENCODING_PCM_16BIT，则为mChannelCount * 2
* mNativeBufferSizeInFrames为帧数
*/
audioBuffSizeCheck(bufferSizeInBytes);
mInitializationLooper = looper;
if (sessionId < 0) {
throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
}
//进入native层初始化
int[] sampleRate = new int[] {mSampleRate};
int[] session = new int[1];
session[0] = sessionId;
// native initialization
int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes,
sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,
mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/,
offload, encapsulationMode, tunerConfiguration);
if (initResult != SUCCESS) {
loge("Error code "+initResult+" when initializing AudioTrack.");
return; // with mState == STATE_UNINITIALIZED
}
mSampleRate = sampleRate[0];
mSessionId = session[0];
// TODO: consider caching encapsulationMode and tunerConfiguration in the Java object.
if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) != 0) {
int frameSizeInBytes;
if (AudioFormat.isEncodingLinearFrames(mAudioFormat)) {
frameSizeInBytes = mChannelCount * AudioFormat.getBytesPerSample(mAudioFormat);
} else {
frameSizeInBytes = 1;
}
mOffset = ((int) Math.ceil(HEADER_V2_SIZE_BYTES / frameSizeInBytes)) * frameSizeInBytes;
}
if (mDataLoadMode == MODE_STATIC) {
mState = STATE_NO_STATIC_DATA;
} else {
mState = STATE_INITIALIZED;
}
baseRegisterPlayer();
}AudioTrack有两种数据载入模式：
MODE_STREAM:上层持续write音频数据流到AudioTrack中，write堵塞直到数据流从Java层传输到native层，同时添加到播放队列中。这中模式适用于播放较大音频数据，但会造成了一定的延时。一般用于播放在线音乐，或是较大的音频文件。
MODE_STATIC：在播放之前，先把全部数据一次性write到AudioTrack的内部缓冲区中。适用于播放内存占用小、延时要求较高的音频数据。一般用于提示音，按键音等.....
AudioTrack的构造中上面做了一些参数的检查，这里的native_setup函数是通过Jni在C++层创建一个AudioTrack。
android_media_AudioTrack.cpp
static const JNINativeMethod gMethods[] = {
// name, signature, funcPtr
{"native_is_direct_output_supported", "(IIIIIII)Z",
(void *)android_media_AudioTrack_is_direct_output_supported},
{"native_start", "()V", (void *)android_media_AudioTrack_start},
{"native_stop", "()V", (void *)android_media_AudioTrack_stop},
{"native_pause", "()V", (void *)android_media_AudioTrack_pause},
{"native_flush", "()V", (void *)android_media_AudioTrack_flush},
{"native_setup", "(Ljava/lang/Object;Ljava/lang/Object;[IIIIII[IJZILjava/lang/Object;)I",
(void *)android_media_AudioTrack_setup},
...
...
}JNI使用的是动态注册的方式，JNI相关的知识这里不做介绍。
static jint android_media_AudioTrack_setup(JNIEnv *env, jobject thiz, jobject weak_this,
jobject jaa, jintArray jSampleRate,
jint channelPositionMask, jint channelIndexMask,
jint audioFormat, jint buffSizeInBytes, jint memoryMode,
jintArray jSession, jlong nativeAudioTrack,
jboolean offload, jint encapsulationMode,
jobject tunerConfiguration) {
...
...
...
// create the native AudioTrack object
lpTrack = new AudioTrack();
// read the AudioAttributes values
auto paa = JNIAudioAttributeHelper::makeUnique();
jint jStatus = JNIAudioAttributeHelper::nativeFromJava(env, jaa, paa.get());
if (jStatus != (jint)AUDIO_JAVA_SUCCESS) {
return jStatus;
}
ALOGV("AudioTrack_setup for usage=%d content=%d flags=0x%#x tags=%s",
paa->usage, paa->content_type, paa->flags, paa->tags);
// initialize the callback information:
// this data will be passed with every AudioTrack callback
// 创建存储音频数据的容器
lpJniStorage = new AudioTrackJniStorage();
//将Java层的AudioTrack引用保存到AudioTrackJniStorage中
lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);
// we use a weak reference so the AudioTrack object can be garbage collected.
lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);
lpJniStorage->mCallbackData.isOffload = offload;
lpJniStorage->mCallbackData.busy = false;
audio_offload_info_t offloadInfo;
if (offload == JNI_TRUE) {
offloadInfo = AUDIO_INFO_INITIALIZER;
offloadInfo.format = format;
offloadInfo.sample_rate = sampleRateInHertz;
offloadInfo.channel_mask = nativeChannelMask;
offloadInfo.has_video = false;
offloadInfo.stream_type = AUDIO_STREAM_MUSIC; //required for offload
}
// initialize the native AudioTrack object
//初始化不同模式下的native AudioTrack对象
status_t status = NO_ERROR;
switch (memoryMode) { //stream模式
case MODE_STREAM:
status = lpTrack->set(
AUDIO_STREAM_DEFAULT,// stream type, but more info conveyed in paa (last argument)
sampleRateInHertz,
format,// word length, PCM
nativeChannelMask,
offload ? 0 : frameCount,
offload ? AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_NONE,
audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user)
0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack
0,//stream模式下的共享内存在AudioFlinger中创建
true,// thread can call Java
sessionId,// audio session ID
offload ? AudioTrack::TRANSFER_SYNC_NOTIF_CALLBACK : AudioTrack::TRANSFER_SYNC,
offload ? &offloadInfo : NULL,
-1, -1, // default uid, pid values
paa.get());
break;
case MODE_STATIC: //static模式
// AudioTrack is using shared memory
// 为AudioTrack分配共享内存区域
if (!lpJniStorage->allocSharedMem(buffSizeInBytes)) {
ALOGE("Error creating AudioTrack in static mode: error creating mem heap base");
goto native_init_failure;
}
status = lpTrack->set(
AUDIO_STREAM_DEFAULT,// stream type, but more info conveyed in paa (last argument)
sampleRateInHertz,
format,// word length, PCM
nativeChannelMask,
frameCount,
AUDIO_OUTPUT_FLAG_NONE,
audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user));
0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack
lpJniStorage->mMemBase,// shared mem
true,// thread can call Java
sessionId,// audio session ID
AudioTrack::TRANSFER_SHARED,
NULL, // default offloadInfo
-1, -1, // default uid, pid values
paa.get());
break;
default:
ALOGE("Unknown mode %d", memoryMode);
goto native_init_failure;
}
if (status != NO_ERROR) {
ALOGE("Error %d initializing AudioTrack", status);
goto native_init_failure;
}
// Set caller name so it can be logged in destructor.
// MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_JAVA
lpTrack->setCallerName("java");
} else { // end if (nativeAudioTrack == 0)
lpTrack = (AudioTrack*)nativeAudioTrack;
// TODO: We need to find out which members of the Java AudioTrack might
// need to be initialized from the Native AudioTrack
// these are directly returned from getters:
// mSampleRate
// mAudioFormat
// mStreamType
// mChannelConfiguration
// mChannelCount
// mState (?)
// mPlayState (?)
// these may be used internally (Java AudioTrack.audioParamCheck():
// mChannelMask
// mChannelIndexMask
// mDataLoadMode
// initialize the callback information:
// this data will be passed with every AudioTrack callback
// 创建存储音频数据的容器
lpJniStorage = new AudioTrackJniStorage();
lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);
// we use a weak reference so the AudioTrack object can be garbage collected.
lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);
lpJniStorage->mCallbackData.busy = false;
}
lpJniStorage->mAudioTrackCallback =
new JNIAudioTrackCallback(env, thiz, lpJniStorage->mCallbackData.audioTrack_ref,
javaAudioTrackFields.postNativeEventInJava);
lpTrack->setAudioTrackCallback(lpJniStorage->mAudioTrackCallback);
nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
if (nSession == NULL) {
ALOGE("Error creating AudioTrack: Error retrieving session id pointer");
goto native_init_failure;
}
// read the audio session ID back from AudioTrack in case we create a new session
...
...
...
...
return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;
}在android_media_AudioTrack_setup()函数中做了几个很重要的事情：
1，创建Native层的Audiorack。
2，创建存储音频数据的容器。
3，将Java层的AudioTrack引用保存到AudioTrackJniStorage中
struct audiotrack_callback_cookie {
jclass audioTrack_class;
jobject audioTrack_ref;
bool busy;
Condition cond;
bool isOffload;
};
class AudioTrackJniStorage {
public:
sp<MemoryHeapBase> mMemHeap;
sp<MemoryBase> mMemBase;
audiotrack_callback_cookie mCallbackData{};
sp<JNIDeviceCallback> mDeviceCallback;
sp<JNIAudioTrackCallback> mAudioTrackCallback;
bool allocSharedMem(int sizeInBytes) {
mMemHeap = new MemoryHeapBase(sizeInBytes, 0, "AudioTrack Heap Base");
if (mMemHeap->getHeapID() < 0) {
return false;
}
mMemBase = new MemoryBase(mMemHeap, 0, sizeInBytes);
return true;
}
};
bool allocSharedMem(int sizeInBytes) {
//创建一个匿名共享内存
mMemHeap = new MemoryHeapBase(sizeInBytes, 0, "AudioTrack Heap Base");
if (mMemHeap->getHeapID() < 0) {
return false;
}
mMemBase = new MemoryBase(mMemHeap, 0, sizeInBytes);
return true;
}
};
//创建匿名共享内存区域
MemoryHeapBase::MemoryHeapBase(size_t size, uint32_t flags, char const * name)
: mFD(-1), mSize(0), mBase(MAP_FAILED), mFlags(flags),
mDevice(nullptr), mNeedUnmap(false), mOffset(0)
{
//获取内存页大小
const size_t pagesize = getpagesize();
//字节对齐
size = ((size + pagesize-1) & ~(pagesize-1));
//创建共享内存。打开/dev/ashmem设备。得到一个文件描写叙述符
int fd = ashmem_create_region(name == nullptr ? "MemoryHeapBase" : name, size);
ALOGE_IF(fd<0, "error creating ashmem region: %s", strerror(errno));
if (fd >= 0) {
//通过mmap将匿名共享内存映射到当前进程地址空间
if (mapfd(fd, size) == NO_ERROR) {
if (flags & READ_ONLY) {
ashmem_set_prot_region(fd, PROT_READ);
}
}
}
}4，初始化不同模式下的native AudioTrack对象，并调用了lpTrack的set()方法。
AudioTrack.cpp
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int32_t notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
uid_t uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed,
audio_port_handle_t selectedDeviceId)
{
status_t status;
uint32_t channelCount;
pid_t callingPid;
pid_t myPid;
...
...
mThreadCanCallJava = threadCanCallJava;
mSelectedDeviceId = selectedDeviceId;
mSessionId = sessionId;
//设置音频传输数据类型
switch (transferType) {
case TRANSFER_DEFAULT:
if (sharedBuffer != 0) {
transferType = TRANSFER_SHARED;
} else if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
case TRANSFER_SYNC_NOTIF_CALLBACK:
if (cbf == NULL || sharedBuffer != 0) {
ALOGE("%s(): Transfer type %s but cbf == NULL || sharedBuffer != 0",
convertTransferToText(transferType), __func__);
status = BAD_VALUE;
goto exit;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("%s(): Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
status = BAD_VALUE;
goto exit;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("%s(): Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
status = BAD_VALUE;
goto exit;
}
break;
default:
ALOGE("%s(): Invalid transfer type %d",
__func__, transferType);
status = BAD_VALUE;
goto exit;
}
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
mDoNotReconnect = doNotReconnect;
ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
__func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
ALOGV("%s(): streamType %d frameCount %zu flags %04x",
__func__, streamType, frameCount, flags);
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("%s(): Track already in use", __func__);
status = INVALID_OPERATION;
goto exit;
}
// handle default values first.
//音频格式设置
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (pAttributes == NULL) {
if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("%s(): Invalid stream type %d", __func__, streamType);
status = BAD_VALUE;
goto exit;
}
mStreamType = streamType;
} else {
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
ALOGV("%s(): Building AudioTrack with attributes:"
" usage=%d content=%d flags=0x%x tags=[%s]",
__func__,
mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
mStreamType = AUDIO_STREAM_DEFAULT;
audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
}
...
...
...
//假设设置了提供音频数据的回调函数，则启动AudioTrackThread线程来提供音频数据
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioTrack
{
AutoMutex lock(mLock);
status = createTrack_l();//继续
}
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
goto exit;
}
...
...
...
mStatus = status;
return status;
}set()函数中对参数做了一些处理，继续看一下createTrack_l()。
status_t AudioTrack::createTrack_l()
{
status_t status;
bool callbackAdded = false;
//得到AudioFlinger的代理对象
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("%s(%d): Could not get audioflinger",
__func__, mPortId);
status = NO_INIT;
goto exit;
}
{
...
...
...
...
IAudioFlinger::CreateTrackInput input;
if (mStreamType != AUDIO_STREAM_DEFAULT) {
input.attr = AudioSystem::streamTypeToAttributes(mStreamType);
} else {
input.attr = mAttributes;
}
input.config = AUDIO_CONFIG_INITIALIZER;
input.config.sample_rate = mSampleRate;
input.config.channel_mask = mChannelMask;
input.config.format = mFormat;
input.config.offload_info = mOffloadInfoCopy;
input.clientInfo.clientUid = mClientUid;
input.clientInfo.clientPid = mClientPid;
input.clientInfo.clientTid = -1;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
// It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
// application-level code follows all non-blocking design rules, the language runtime
// doesn't also follow those rules, so the thread will not benefit overall.
if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
input.clientInfo.clientTid = mAudioTrackThread->getTid();
}
}
input.sharedBuffer = mSharedBuffer;
input.notificationsPerBuffer = mNotificationsPerBufferReq;
input.speed = 1.0;
if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
(mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
}
input.flags = mFlags;
input.frameCount = mReqFrameCount;
input.notificationFrameCount = mNotificationFramesReq;
input.selectedDeviceId = mSelectedDeviceId;
input.sessionId = mSessionId;
input.audioTrackCallback = mAudioTrackCallback;
IAudioFlinger::CreateTrackOutput output;
sp<IAudioTrack> track = audioFlinger->createTrack(input,
output,
&status);
...\
...
...
...
...
// FIXME compare to AudioRecord
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
status = NO_INIT;
goto exit;
}
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
void *iMemPointer = iMem->unsecurePointer();
if (iMemPointer == NULL) {
ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
status = NO_INIT;
goto exit;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
//将创建的Track代理对象、匿名共享内存代理对象保存到AudioTrack的成员变量中
mAudioTrack = track;
mCblkMemory = iMem;
IPCThreadState::self()->flushCommands();
//保存匿名共享内存的首地址。在匿名共享内存的头部存放了一个audio_track_cblk_t对象
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
__func__, mPortId, mReqFrameCount, mFrameCount);
if (!mThreadCanCallJava) {
mAwaitBoost = true;
}
} else {
ALOGD("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
__func__, mPortId, mReqFrameCount, mFrameCount);
}
}
mFlags = output.flags;
//mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
if (mDeviceCallback != 0) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
}
AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
callbackAdded = true;
}
mPortId = output.portId;
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output.outputId;
mRefreshRemaining = true;
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
// is the value of pointer() for the shared buffer, otherwise buffers points
// immediately after the control block. This address is for the mapping within client
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
if (mSharedBuffer == 0) {
buffers = cblk + 1;
} else {
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
buffers = mSharedBuffer->unsecurePointer();
if (buffers == NULL) {
ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
status = NO_INIT;
goto exit;
}
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (mFrameCount > mReqFrameCount) {
mReqFrameCount = mFrameCount;
}
// reset server position to 0 as we have new cblk.
mServer = 0;
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy = mStaticProxy;
}
mProxy->setVolumeLR(gain_minifloat_pack(
gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
mProxy->setSendLevel(mSendLevel);
const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
mProxy->setSampleRate(effectiveSampleRate);
AudioPlaybackRate playbackRateTemp = mPlaybackRate;
playbackRateTemp.mSpeed = effectiveSpeed;
playbackRateTemp.mPitch = effectivePitch;
mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setMinimum(mNotificationFramesAct);
mDeathNotifier = new DeathNotifier(this);
IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
// This is the first log sent from the AudioTrack client.
// The creation of the audio track by AudioFlinger (in the code above)
// is the first log of the AudioTrack and must be present before
// any AudioTrack client logs will be accepted.
...
...
...
...
...
mStatus = status;
// sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
return status;
}上一篇文章中讲到，AudioPolicyService初始化的时候创建了AudioPolicyManager并调用了AudioFlinger::openOutput()函数打开一个默认的音频输出PlaybackThread线程，同时为该线程分配一个全局唯一的audio_io_handle_t值，并以键值对的形式保存在AudioFlinger的成员变量mPlaybackThreads中。在这里首先依据音频參数通过调用AudioSystem::getOutput()函数得到当前音频输出接口的PlaybackThread线程id号。同时传递给createTrack函数用于创建Track。
音频播放需要AudioTrack写入音频数据，同时AudioFlinger拿到数据后做混音，因此要在AudioTrack与AudioFlinger之间建立数据通道，而AudioTrack与AudioFlinger又分属不同的进程空间，Android系统採用Binder通信方式来通信。
IAudioTrack建立了AudioTrack与AudioFlinger之间的关系，在static模式下，用于存放音频数据的匿名共享内存在AudioTrack这边创建。在stream播放模式下，匿名共享内存却是在AudioFlinger这边创建。这两种播放模式下创建的匿名共享内存是有差别的，stream模式下的匿名共享内存头部会创建一个audio_track_cblk_t对象，用于协调生产者AudioTrack和消费者AudioFlinger之间的步调。createTrack就是在AudioFlinger中创建一个Track对象。
AudioFlinger.cpp
sp<IAudioTrack> AudioFlinger::createTrack(const CreateTrackInput& input,
CreateTrackOutput& output,
status_t *status)
{
sp<PlaybackThread::Track> track; //创建一个track对象
sp<TrackHandle> trackHandle;
sp<Client> client;
status_t lStatus;
audio_stream_type_t streamType;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
std::vector<audio_io_handle_t> secondaryOutputs;
bool updatePid = (input.clientInfo.clientPid == -1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
uid_t clientUid = input.clientInfo.clientUid;
audio_io_handle_t effectThreadId = AUDIO_IO_HANDLE_NONE;
std::vector<int> effectIds;
audio_attributes_t localAttr = input.attr;
if (!isAudioServerOrMediaServerUid(callingUid)) {
ALOGW_IF(clientUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, clientUid);
clientUid = callingUid;
updatePid = true;
}
pid_t clientPid = input.clientInfo.clientPid;
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
if (updatePid) {
ALOGW_IF(clientPid != -1 && clientPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, clientPid);
clientPid = callingPid;
}
audio_session_t sessionId = input.sessionId;
if (sessionId == AUDIO_SESSION_ALLOCATE) {
sessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
} else if (audio_unique_id_get_use(sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
lStatus = BAD_VALUE;
goto Exit;
}
output.sessionId = sessionId;
output.outputId = AUDIO_IO_HANDLE_NONE;
output.selectedDeviceId = input.selectedDeviceId;
//获取输出设备（这个函数最终会在AudioPolicyManager.cpp中实现，通过mEngine直接获取device）
lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
clientPid, clientUid, &input.config, input.flags,
&output.selectedDeviceId, &portId, &secondaryOutputs);
if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
goto Exit;
}
....
....
....
....
....
{
Mutex::Autolock _l(mLock);
//依据播放线程ID号查找出相应的PlaybackThread，在openout时，播放线程以key/value形式保存在AudioFlinger的mPlaybackThreads中
PlaybackThread *thread = checkPlaybackThread_l(output.outputId);
if (thread == NULL) {
ALOGE("no playback thread found for output handle %d", output.outputId);
lStatus = BAD_VALUE;
goto Exit;
}
//依据客户端进程pid查找是否已经为该客户进程创建了Client对象。假设没有，则创建一个Client对象
client = registerPid(clientPid);
PlaybackThread *effectThread = NULL;
// check if an effect chain with the same session ID is present on another
// output thread and move it here.
//遍历全部的播放线程。不包含输出线程，假设该线程中Track的sessionId与当前同样，则取出该线程作为当前Track的effectThread。
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output.outputId) {
uint32_t sessions = t->hasAudioSession(sessionId);
if (sessions & ThreadBase::EFFECT_SESSION) {
effectThread = t.get();
break;
}
}
}
ALOGV("createTrack() sessionId: %d", sessionId);
output.sampleRate = input.config.sample_rate;
output.frameCount = input.frameCount;
output.notificationFrameCount = input.notificationFrameCount;
output.flags = input.flags;
//在找到的PlaybackThread线程中创建Track
track = thread->createTrack_l(client, streamType, localAttr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, &output.notificationFrameCount,
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
callingPid, input.clientInfo.clientTid, clientUid,
&lStatus, portId, input.audioTrackCallback);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
output.afFrameCount = thread->frameCount();
output.afSampleRate = thread->sampleRate();
output.afLatencyMs = thread->latency();
output.portId = portId;
if (lStatus == NO_ERROR) {
// Connect secondary outputs. Failure on a secondary output must not imped the primary
// Any secondary output setup failure will lead to a desync between the AP and AF until
// the track is destroyed.
TeePatches teePatches;
for (audio_io_handle_t secondaryOutput : secondaryOutputs) {
PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
if (secondaryThread == NULL) {
ALOGE("no playback thread found for secondary output %d", output.outputId);
continue;
}
....
....
....
....
....
....
//此时Track已成功创建，还须要为该Track创建代理对象TrackHandle
// return handle to client
trackHandle = new TrackHandle(track);
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseOutput(portId);
}
*status = lStatus;
return trackHandle;
}AudioFlinger中为应用程序进程创建一个Client对象，与客户进程通信。然后跟据播放线程ID找出相应的PlaybackThread。并将创建Track的任务转交给它，PlaybackThread完毕Track创建后，因为Track没有通信功能，因此还须要为其创建一个代理通信业务的TrackHandle对象。
(网上资源)
//依据进程pid。为请求播放音频的client创建一个Client对象。
sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
{
Mutex::Autolock _cl(mClientLock);
// If pid is already in the mClients wp<> map, then use that entry
// (for which promote() is always != 0), otherwise create a new entry and Client.
sp<Client> client = mClients.valueFor(pid).promote();
if (client == 0) {
client = new Client(this, pid);
mClients.add(pid, client);
}
return client;
}
AudioFlinger的变量mClients以键值对的形式保存pid和Client对象。这里首先取出pid相应的Client对象，如果没有，则为client进程创建一个新的Client对象。
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mPid(pid)
{
mMemoryDealer = new MemoryDealer(
audioFlinger->getClientSharedHeapSize(),
(std::string("AudioFlinger::Client(") + std::to_string(pid) + ")").c_str());
}创建了一个MemoryDealer对象，该对象用于分配共享内存
MemoryDealer::MemoryDealer(size_t size, const char* name, uint32_t flags)
: mHeap(new MemoryHeapBase(size, flags, name)), //创建指定大小的共享内存
mAllocator(new SimpleBestFitAllocator(size)) //创建内存分配器
{
}
SimpleBestFitAllocator::SimpleBestFitAllocator(size_t size)
{
size_t pagesize = getpagesize();
mHeapSize = ((size + pagesize-1) & ~(pagesize-1));
chunk_t* node = new chunk_t(0, mHeapSize / kMemoryAlign);
mList.insertHead(node);
}当应用程序进程中的AudioTrack请求AudioFlinger在PlaybackThread中创建Track对象时，AudioFlinger首先会为应用程序进程创建一个Client对象，同一时候创建一块大小为2M的共享内存。在创建Track时，Track将在2M共享内存中分配buffer用于音频播放。
Threads.cpp
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
size_t *pNotificationFrameCount,
uint32_t notificationsPerBuffer,
float speed,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
audio_output_flags_t *flags,
pid_t creatorPid,
pid_t tid,
uid_t uid,
status_t *status,
audio_port_handle_t portId,
const sp<media::IAudioTrackCallback>& callback)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
sp<Track> track;
status_t lStatus;
audio_output_flags_t outputFlags = mOutput->flags;
audio_output_flags_t requestedFlags = *flags;
uint32_t sampleRate;
if (sharedBuffer != 0 && checkIMemory(sharedBuffer) != NO_ERROR) {
lStatus = BAD_VALUE;
goto Exit;
}
if (*pSampleRate == 0) {
*pSampleRate = mSampleRate;
}
sampleRate = *pSampleRate;
// special case for FAST flag considered OK if fast mixer is present
if (hasFastMixer()) {
outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST);
}
....
....
....
....
....
track = new Track(this, client, streamType, attr, sampleRate, format,
channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
sessionId, creatorPid, uid, *flags, TrackBase::TYPE_DEFAULT, portId);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
// track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
{
Mutex::Autolock _atCbL(mAudioTrackCbLock);
if (callback.get() != nullptr) {
mAudioTrackCallbacks.emplace(callback);
}
}
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) {
pid_t callingPid = IPCThreadState::self()->getCallingPid();
// we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
// so ask activity manager to do this on our behalf
sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/);
}
}
lStatus = NO_ERROR;
Exit:
*status = lStatus;
return track;
}上面函数中创建了一个Track，而Track继承了TrackBase所以我们看下它的构造
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
ThreadBase *thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_session_t sessionId,
pid_t creatorPid,
uid_t clientUid,
bool isOut,
alloc_type alloc,
track_type type,
audio_port_handle_t portId,
std::string metricsId)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(NULL),
// mBuffer, mBufferSize
mState(IDLE),
mAttr(attr),
mSampleRate(sampleRate),
mFormat(format),
mChannelMask(channelMask),
mChannelCount(isOut ?
audio_channel_count_from_out_mask(channelMask) :
audio_channel_count_from_in_mask(channelMask)),
mFrameSize(audio_has_proportional_frames(format) ?
mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
mFrameCount(frameCount),
mSessionId(sessionId),
mIsOut(isOut),
mId(android_atomic_inc(&nextTrackId)),
mTerminated(false),
mType(type),
mThreadIoHandle(thread ? thread->id() : AUDIO_IO_HANDLE_NONE),
mPortId(portId),
mIsInvalid(false),
mTrackMetrics(std::move(metricsId), isOut),
mCreatorPid(creatorPid)
{
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
if (!isAudioServerOrMediaServerUid(callingUid) || clientUid == AUDIO_UID_INVALID) {
ALOGW_IF(clientUid != AUDIO_UID_INVALID && clientUid != callingUid,
"%s(%d): uid %d tried to pass itself off as %d",
__func__, mId, callingUid, clientUid);
clientUid = callingUid;
}
// clientUid contains the uid of the app that is responsible for this track, so we can blame
// battery usage on it.
//得到应用进程uid
mUid = clientUid;
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount;
// check overflow when computing bufferSize due to multiplication by mFrameSize.
if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2
|| mFrameSize == 0 // format needs to be correct
|| minBufferSize > SIZE_MAX / mFrameSize) {
android_errorWriteLog(0x534e4554, "34749571");
return;
}
minBufferSize *= mFrameSize;
if (buffer == nullptr) {
bufferSize = minBufferSize; // allocated here.
} else if (minBufferSize > bufferSize) {
android_errorWriteLog(0x534e4554, "38340117");
return;
}
size_t size = sizeof(audio_track_cblk_t);
if (buffer == NULL && alloc == ALLOC_CBLK) {
// check overflow when computing allocation size for streaming tracks.
if (size > SIZE_MAX - bufferSize) {
android_errorWriteLog(0x534e4554, "34749571");
return;
}
size += bufferSize;
}
if (client != 0) {
//请求Client中的MemoryDealer工具类来分配buffer
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory == 0 ||
(mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->unsecurePointer())) == NULL) {
ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
client->heap()->dump("AudioTrack");
mCblkMemory.clear();
return;
}
} else {
mCblk = (audio_track_cblk_t *) malloc(size);
if (mCblk == NULL) {
ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
return;
}
}
....
....
....
....
mBufferSize = bufferSize;
#ifdef TEE_SINK
mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK);
#endif
}
}TrackBase构造过程主要是为音频播放分配共享内存,继续分析一下Track的构造。
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
...
...
...
mVolumeHandler(new media::VolumeHandler(sampleRate)),
mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(uid, attr, id(), streamType)),
// mSinkTimestamp
mFrameCountToBeReady(frameCountToBeReady),
mFastIndex(-1),
mCachedVolume(1.0),
/* The track might not play immediately after being active, similarly as if its volume was 0.
* When the track starts playing, its volume will be computed. */
mFinalVolume(0.f),
mResumeToStopping(false),
mFlushHwPending(false),
mFlags(flags)
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
ALOGV_IF(sharedBuffer != 0, "%s(%d): sharedBuffer: %p, size: %zu",
__func__, mId, sharedBuffer->unsecurePointer(), sharedBuffer->size());
if (mCblk == NULL) {
return;
}
if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
ALOGE("%s(%d): no more tracks available", __func__, mId);
releaseCblk(); // this makes the track invalid.
return;
}
if (sharedBuffer == 0) { //stream模式
mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack(), sampleRate);
} else { //static模式
mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, sampleRate);
}
mServerProxy = mAudioTrackServerProxy;
...
...
...
...
}在Track构造中。依据不同的播放模式，创建不同的代理对象。
在构造Client对象时，创建了一个内存分配工具对象MemoryDealer，同一时候创建了一块大小为2M的匿名共享内存。这里就是使用MemoryDealer对象在这块匿名共享内存上分配指定大小的buffer。
sp<IMemory> MemoryDealer::allocate(size_t size)
{
sp<IMemory> memory;
//分配size大小的共享内存，并返回该buffer的偏移量
const ssize_t offset = allocator()->allocate(size);
if (offset >= 0) {
将分配的buffer封装为Allocation对象
memory = new Allocation(this, heap(), offset, size);
}
return memory;
}
size_t SimpleBestFitAllocator::allocate(size_t size, uint32_t flags)
{
Mutex::Autolock _l(mLock);
ssize_t offset = alloc(size, flags);
return offset;
}
ssize_t SimpleBestFitAllocator::alloc(size_t size, uint32_t flags)
{
if (size == 0) {
return 0;
}
size = (size + kMemoryAlign-1) / kMemoryAlign;
chunk_t* free_chunk = nullptr;
chunk_t* cur = mList.head();
size_t pagesize = getpagesize();
while (cur) {
int extra = 0;
if (flags & PAGE_ALIGNED)
extra = ( -cur->start & ((pagesize/kMemoryAlign)-1) ) ;
// best fit
if (cur->free && (cur->size >= (size+extra))) {
if ((!free_chunk) || (cur->size < free_chunk->size)) {
free_chunk = cur;
}
if (cur->size == size) {
break;
}
}
cur = cur->next;
}
if (free_chunk) {
const size_t free_size = free_chunk->size;
free_chunk->free = 0;
free_chunk->size = size;
if (free_size > size) {
int extra = 0;
if (flags & PAGE_ALIGNED)
extra = ( -free_chunk->start & ((pagesize/kMemoryAlign)-1) ) ;
if (extra) {
chunk_t* split = new chunk_t(free_chunk->start, extra);
free_chunk->start += extra;
mList.insertBefore(free_chunk, split);
}
ALOGE_IF((flags&PAGE_ALIGNED) &&
((free_chunk->start*kMemoryAlign)&(pagesize-1)),
"PAGE_ALIGNED requested, but page is not aligned!!!");
const ssize_t tail_free = free_size - (size+extra);
if (tail_free > 0) {
chunk_t* split = new chunk_t(
free_chunk->start + free_chunk->size, tail_free);
mList.insertAfter(free_chunk, split);
}
}
return (free_chunk->start)*kMemoryAlign;
}
return NO_MEMORY;
}
createTrack时由AudioFlinger申请对应的内存，然后通过IMemory接口返回AudioTrack，这样AudioTrack和AudioFlinger管理着同一个audio_track_cblk_t，通过它实现了环形数据共享区。AudioTrack向数据共享区中写入音频数据，AudioFlinger从数据共享区中读取音频数据，经Mixer后送给AudioHardware进行播放。这里AudioTrack是生产者，AudioFlinger是消费者。
总结一下放音的流程：
1、Java层通过JNI创建AudioTrack对象。
2、根据StreamType等參数，查找已打开的音频输出设备。如果查找不到匹配的音频输出设备。则请求AudioFlinger打开新的音频输出设备
3、AudioFlinger为该输出设备创建混音线程MixerThread，并把该线程的id作为getOutput()的返回值返回给AudioTrack;
4、AudioTrack通过binder机制调用AudioFlinger的createTrack()创建Track。而且创建TrackHandle Binder本地对象，同时返回IAudioTrack的代理对象。
5、AudioFlinger注冊该Track到MixerThread中。
6、AudioTrack通过IAudioTrack接口，得到在AudioFlinger中创建的数据共享区。
7、AudioTrack调用start，开始写入数据，AudioFlinger读取数据，进行放音。
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