基于Android 9.0
1.概述
主要是对Native层AudioTrack的构造器进行了详细的分析。从前一章节《Android 音频子系统--07：AudioTrack实例详解》分析可以知道，Java层AudioTrack最终也是调用Native层的AudioTrack，所以我们分析的核心就是Native层的AudioTrack。本章节主要分析AudioTrack是如何跟output和playbackthread建立关联的。
2.AudioTrack构造器分析
C++层的AudioTrack对象它的构造器代码如下：
// frameworks/av/media/libaudioclient/AudioTrack.cpp
//无参构造方法，后再调用set方法
AudioTrack::AudioTrack()
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
mAttributes.flags = 0x0;
strcpy(mAttributes.tags, "");
}
// ---------------
//有参对象，不需再调用set方法
AudioTrack::AudioTrack(
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,
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)
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
// 构造方法里面自动调用set()方法
(void)set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
}
可以看到 AudioTrack的使用有两种情况：
- 无参数构造器，后期需要再执行set方法设置参数。
- 有参数构造器，会直接调用set方法设置参数。
3.set函数详细分析
接下来对AudioTrack的set方法（核心方法）进行分析，set函数的代码如下：
// frameworks/av/media/libaudioclient/AudioTrack.cpp
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
// ......
audio_port_handle_t selectedDeviceId)
{
//......
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;
}
//设置音频数据传输类型..
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
mDoNotReconnect = doNotReconnect;
// ......
// handle default values first.
//音频流类型设置,程序会设为默认值AUDIO_STREAM_MUSIC
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;
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
mAttributes.flags = 0x0;
strcpy(mAttributes.tags, "");
} 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);
}
// these below should probably come from the audioFlinger too...
//音频格式设置,采样深度默认为16bit
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
} else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("%s(): Invalid format %#x", __func__, format);
status = BAD_VALUE;
goto exit;
}
mFormat = format;
//输出声道合法性检查
if (!audio_is_output_channel(channelMask)) {
ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
status = BAD_VALUE;
goto exit;
}
mChannelMask = channelMask;
channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
// ......
//根据音频流类型从AudioPolicyService中得到对应的音频采样率
// sampling rate must be specified for direct outputs
if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
status = BAD_VALUE;
goto exit;
}
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
// Make copy of input parameter offloadInfo so that in the future:
// (a) createTrack_l doesn't need it as an input parameter
// (b) we can support re-creation of offloaded tracks
if (offloadInfo != NULL) {
mOffloadInfoCopy = *offloadInfo;
mOffloadInfo = &mOffloadInfoCopy;
} else {
mOffloadInfo = NULL;
memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
}
//左右声道初始音量都设置成最大
mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
mSendLevel = 0.0f;
// mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
if (notificationFrames >= 0) {
mNotificationFramesReq = notificationFrames;
mNotificationsPerBufferReq = 0;
} else {
if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
ALOGE("%s(): notificationFrames=%d not permitted for non-fast track",
__func__, notificationFrames);
status = BAD_VALUE;
goto exit;
}
if (frameCount > 0) {
ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
__func__, notificationFrames, frameCount);
status = BAD_VALUE;
goto exit;
}
mNotificationFramesReq = 0;
const uint32_t minNotificationsPerBuffer = 1;
const uint32_t maxNotificationsPerBuffer = 8;
mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
"%s(): notificationFrames=%d clamped to the range -%u to -%u",
__func__,
notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
}
mNotificationFramesAct = 0;
callingPid = IPCThreadState::self()->getCallingPid();
myPid = getpid();
if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingPid != myPid)) {
mClientPid = callingPid;
} else {
mClientPid = pid;
}
mAuxEffectId = 0;
mOrigFlags = mFlags = flags;
mCbf = cbf;
//如果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) {
/*AudioTrackThread实现两个核心功能:
*1 AudioTrack与AudioFlinger之间数据传输，AudioFlinger启动了一个线程专门用于接收客户端的
* 音频数据，同时客户端也需要一个线程来“不断”的传送音频数据
*2 用于报告数据的传输状态,AudioTrack中保存了一个callback_t类型的回调函数（即全局变量mCbf）
* 用于事件发生时进行回传
*/
mAudioTrackThread = new AudioTrackThread(*this);
//运行线程
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();
}
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
goto exit;
}
// ......
exit:
mStatus = status;
return status;
}
这里专注于分析createTrack_l方法，代码实现如下：
// frameworks/av/media/libaudioclient/AudioTrack.cpp
status_t AudioTrack::createTrack_l()
{
status_t status;
bool callbackAdded = false;
// 获取AudioFlinger
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
// ......
// streamType映射成Attributes，后续都是Attributes处理
IAudioFlinger::CreateTrackInput input;
if (mStreamType != AUDIO_STREAM_DEFAULT) {
input.attr = AudioSystem::streamTypeToAttributes(mStreamType);
} else {
input.attr = mAttributes;
}
// 省略：input配置参数
IAudioFlinger::CreateTrackOutput output;
/*创建Track，这里返回AudioFlinger内部的AudioTrack的binder代理Track
*它是联系AudioTrack和AudioFlinger的关键纽带
*/
sp<IAudioTrack> track = audioFlinger->createTrack(input,
output,
&status);
// ......
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
// FIXME compare to AudioRecord
/*获取 track变量 的共享内存 buffer
*当PlaybackThread创建一个PlaybackThread::Track对象时，所需的缓冲区空间
*就已经分配了，这块空间是可以跨进程共享的，所以AudioTrack可以通过track->getCblk
*来获取共享内存了。
*/
sp<IMemory> iMem = track->getCblk();//向AudioFlinger申请数据缓冲空间
if (iMem == 0) {
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
status = NO_INIT;
goto exit;
}
void *iMemPointer = iMem->pointer();
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();
}
//与AudioFlinger进行通信的中介
mAudioTrack = track;
mCblkMemory = iMem;
IPCThreadState::self()->flushCommands();
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
//...
// update proxy，APP的AudioTack 与Thread中的Track建立共享内存
// Proxy类封装了 track 共享 buffer 的操控接口,实现共享 buffer 的使用
// 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 {
buffers = mSharedBuffer->pointer();
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;
}
//调用 Proxy 的 set 接口,设置保存 VolumeLR,SampleRate,SendLevel 等参数,
//AudioFlinger mixer 线程中会把这些参数取出来实现混音
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);
}
exit:
if (status != NO_ERROR && callbackAdded) {
// note: mOutput is always valid is callbackAdded is true
AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
}
mStatus = status;
// sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
return status;
}
3.1 AudioFlinger的createTrack方法
// frameworks/av/services/audioflinger/AudioFlinger.cpp
sp<IAudioTrack> AudioFlinger::createTrack(const CreateTrackInput& input,
CreateTrackOutput& output,
status_t *status)
{
// ......
output.sessionId = sessionId;
output.outputId = AUDIO_IO_HANDLE_NONE;
output.selectedDeviceId = input.selectedDeviceId;
//关键点1：通过属性获取output。
lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
clientPid, clientUid, &input.config, input.flags,
&output.selectedDeviceId, &portId, &secondaryOutputs);
// 省略：检查一些参数是否合格
{
Mutex::Autolock _l(mLock);
// 创建PlaybackThread线程
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;
}
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.
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;
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);
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;
}
size_t sourceFrameCount = thread->frameCount() * output.sampleRate
/ thread->sampleRate();
size_t sinkFrameCount = secondaryThread->frameCount() * output.sampleRate
/ secondaryThread->sampleRate();
// If the secondary output has just been opened, the first secondaryThread write
// will not block as it will fill the empty startup buffer of the HAL,
// so a second sink buffer needs to be ready for the immediate next blocking write.
// Additionally, have a margin of one main thread buffer as the scheduling jitter
// can reorder the writes (eg if thread A&B have the same write intervale,
// the scheduler could schedule AB...BA)
size_t frameCountToBeReady = 2 * sinkFrameCount + sourceFrameCount;
// Total secondary output buffer must be at least as the read frames plus
// the margin of a few buffers on both sides in case the
// threads scheduling has some jitter.
// That value should not impact latency as the secondary track is started before
// its buffer is full, see frameCountToBeReady.
size_t frameCount = frameCountToBeReady + 2 * (sourceFrameCount + sinkFrameCount);
// The frameCount should also not be smaller than the secondary thread min frame
// count
size_t minFrameCount = AudioSystem::calculateMinFrameCount(
[&] { Mutex::Autolock _l(secondaryThread->mLock);
return secondaryThread->latency_l(); }(),
secondaryThread->mNormalFrameCount,
secondaryThread->mSampleRate,
output.sampleRate,
input.speed);
frameCount = std::max(frameCount, minFrameCount);
using namespace std::chrono_literals;
auto inChannelMask = audio_channel_mask_out_to_in(input.config.channel_mask);
sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,
output.sampleRate,
inChannelMask,
input.config.format,
frameCount,
NULL /* buffer */,
(size_t)0 /* bufferSize */,
AUDIO_INPUT_FLAG_DIRECT,
0ns /* timeout */);
status_t status = patchRecord->initCheck();
if (status != NO_ERROR) {
ALOGE("Secondary output patchRecord init failed: %d", status);
continue;
}
// TODO: We could check compatibility of the secondaryThread with the PatchTrack
// for fast usage: thread has fast mixer, sample rate matches, etc.;
// for now, we exclude fast tracks by removing the Fast flag.
const audio_output_flags_t outputFlags =
(audio_output_flags_t)(output.flags & ~AUDIO_OUTPUT_FLAG_FAST);
sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,
streamType,
output.sampleRate,
input.config.channel_mask,
input.config.format,
frameCount,
patchRecord->buffer(),
patchRecord->bufferSize(),
outputFlags,
0ns /* timeout */,
frameCountToBeReady);
status = patchTrack->initCheck();
if (status != NO_ERROR) {
ALOGE("Secondary output patchTrack init failed: %d", status);
continue;
}
teePatches.push_back({patchRecord, patchTrack});
secondaryThread->addPatchTrack(patchTrack);
// In case the downstream patchTrack on the secondaryThread temporarily outlives
// our created track, ensure the corresponding patchRecord is still alive.
patchTrack->setPeerProxy(patchRecord, true /* holdReference */);
patchRecord->setPeerProxy(patchTrack, false /* holdReference */);
}
track->setTeePatches(std::move(teePatches));
}
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
// no risk of deadlock because AudioFlinger::mLock is held
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
if (moveEffectChain_l(sessionId, effectThread, thread) == NO_ERROR) {
effectThreadId = thread->id();
effectIds = thread->getEffectIds_l(sessionId);
}
}
// Look for sync events awaiting for a session to be used.
for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
if (mPendingSyncEvents[i]->triggerSession() == sessionId) {
if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
if (lStatus == NO_ERROR) {
(void) track->setSyncEvent(mPendingSyncEvents[i]);
} else {
mPendingSyncEvents[i]->cancel();
}
mPendingSyncEvents.removeAt(i);
i--;
}
}
}
setAudioHwSyncForSession_l(thread, sessionId);
}
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the Track so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
// Don't hold mClientLock when releasing the reference on the track as the
// destructor will acquire it.
{
Mutex::Autolock _cl(mClientLock);
client.clear();
}
track.clear();
goto Exit;
}
// effectThreadId is not NONE if an effect chain corresponding to the track session
// was found on another thread and must be moved on this thread
if (effectThreadId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::moveEffectsToIo(effectIds, effectThreadId);
}
// 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;
}
3.2 getOutputForAttr
这里专注分析 AudioSystem::getOutputForAttr的实现，代码如下：
status_t AudioSystem::getOutputForAttr(audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
pid_t pid,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return NO_INIT;
return aps->getOutputForAttr(attr, output, session, stream, pid, uid,
config,
flags, selectedDeviceId, portId, secondaryOutputs);
}
这里主要调用了AudioPolicyManager的getOutputForAttr，代码实现如下：
status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs)
{
// ......
// 调用getOutputForAttrInt
status_t status = getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid,
config, flags, selectedDeviceId, &isRequestedDeviceForExclusiveUse,
&secondaryOutputDescs);
if (status != NO_ERROR) {
return status;
}
std::vector<wp<SwAudioOutputDescriptor>> weakSecondaryOutputDescs;
for (auto& secondaryDesc : secondaryOutputDescs) {
secondaryOutputs->push_back(secondaryDesc->mIoHandle);
weakSecondaryOutputDescs.push_back(secondaryDesc);
}
audio_config_base_t clientConfig = {.sample_rate = config->sample_rate,
.format = config->format,
.channel_mask = config->channel_mask };
*portId = AudioPort::getNextUniqueId();
sp<TrackClientDescriptor> clientDesc =
new TrackClientDescriptor(*portId, uid, session, resultAttr, clientConfig,
sanitizedRequestedPortId, *stream,
mEngine->getProductStrategyForAttributes(resultAttr),
toVolumeSource(resultAttr),
*flags, isRequestedDeviceForExclusiveUse,
std::move(weakSecondaryOutputDescs));
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(*output);
outputDesc->addClient(clientDesc);
ALOGV("%s() returns output %d requestedPortId %d selectedDeviceId %d for port ID %d", __func__,
*output, requestedPortId, *selectedDeviceId, *portId);
return NO_ERROR;
}
继续看一下getOutputForAttrInt()方法，代码实现如下：
status_t AudioPolicyManager::getOutputForAttrInt(
audio_attributes_t *resultAttr,
audio_io_handle_t *output,
audio_session_t session,
const audio_attributes_t *attr,
audio_stream_type_t *stream,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
bool *isRequestedDeviceForExclusiveUse,
std::vector<sp<SwAudioOutputDescriptor>> *secondaryDescs)
{
// ......
// 获取音频的Attributes属性
status_t status = getAudioAttributes(resultAttr, attr, *stream);
if (status != NO_ERROR) {
return status;
}
// ......
// explicit routing managed by getDeviceForStrategy in APM is now handled by engine
// in order to let the choice of the order to future vendor engine
// 根据Attributes属性获取OutputDevices
outputDevices = mEngine->getOutputDevicesForAttributes(*resultAttr, requestedDevice, false);
// ......
if (*output == AUDIO_IO_HANDLE_NONE) {
// 根据Devices获取Output
*output = getOutputForDevices(outputDevices, session, *stream, config,
flags, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
}
if (*output == AUDIO_IO_HANDLE_NONE) {
return INVALID_OPERATION;
}
// 选择devices
*selectedDeviceId = getFirstDeviceId(outputDevices);
ALOGV("%s returns output %d selectedDeviceId %d", __func__, *output, *selectedDeviceId);
return NO_ERROR;
}
先看一下getOutputDevicesForAttributes()方法：
DeviceVector Engine::getOutputDevicesForAttributes(const audio_attributes_t &attributes,
const sp<DeviceDescriptor> &preferredDevice,
bool fromCache) const
{
// First check for explict routing device
if (preferredDevice != nullptr) {
ALOGV("%s explicit Routing on device %s", __func__, preferredDevice->toString().c_str());
return DeviceVector(preferredDevice);
}
product_strategy_t strategy = getProductStrategyForAttributes(attributes);
const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
//
// @TODO: what is the priority of explicit routing? Shall it be considered first as it used to
// be by APM?
//
// Honor explicit routing requests only if all active clients have a preferred route in which
// case the last active client route is used
sp<DeviceDescriptor> device = findPreferredDevice(outputs, strategy, availableOutputDevices);
if (device != nullptr) {
return DeviceVector(device);
}
return fromCache? mDevicesForStrategies.at(strategy) : getDevicesForProductStrategy(strategy);
}
3.3 创建PlaybackThread
再回到AudioFlinger的createTrack方法中，会去创建播放线程。
{
Mutex::Autolock _l(mLock);
// 创建PlaybackThread线程
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;
}
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.
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;
// 根据线程来执行createTrack_l来创建一个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);
专注分析checkPlaybackThread_l，代码实现如下：
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
return mPlaybackThreads.valueFor(output).get();
}
这里就是通过output来获取对应的PlaybackThread类型的线程。
3.4 createTrack_l分析
接下来 专注分析createTrack_l，代码实现如下：
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
//...
int uid,
status_t *status)
{
size_t frameCount = *pFrameCount;
sp<Track> track;
status_t lStatus;
bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0;
//...
if (!isTimed) {
//创建新的Track
track = new Track(this, client, streamType, sampleRate, format,
channelMask, frameCount, NULL, sharedBuffer,
sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);
} else {
track = TimedTrack::create(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId, uid);
}
// new Track always returns non-NULL,
// but TimedTrack::create() is a factory that could fail by returning NULL
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;
}
//将track加入到PlayBackThread的mTracks表中。
mTracks.add(track);
//...
lStatus = NO_ERROR;
Exit:
*status = lStatus;
return track;
}
这里主要是playbackthread创建新的track并把track加入到自己的mTracks表中。
