先上一下上一篇 结尾 AudioTrack::createTrack_l 中重要的代码
// 1 重要代码 获取 AudioFlinger
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
// 重要代码2
//调用 audioFlinger 创建Track
sp<IAudioTrack> track = audioFlinger->createTrack(input, output, &status);
// 重要代码3
// 关联buffer 和 内存,向 Hal 层写数据
if (mSharedBuffer == 0) {//注意这里0代表的是Stream
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy = mStaticProxy;
}
看到 使用 Binder 的机制获取了AudioFlinger,利用 AudioFlinger 获取了共享内存的地址,同时创建一个 track,并且将 buffer 和 内存 关联起来,为后续写数据做准备
接下来分析一下 AudioFlinger::createTrack 方法
sp<IAudioTrack> AudioFlinger::createTrack 方法(const CreateTrackInput& input,
CreateTrackOutput& output,
status_t *status)
{
sp<PlaybackThread::Track> track;
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;
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;
}
// client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
// but if someone uses binder directly they could bypass that and cause us to crash
if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
ALOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
// further channel mask checks are performed by createTrack_l() depending on the thread type
if (!audio_is_output_channel(input.config.channel_mask)) {
ALOGE("createTrack() invalid channel mask %#x", input.config.channel_mask);
lStatus = BAD_VALUE;
goto Exit;
}
// further format checks are performed by createTrack_l() depending on the thread type
if (!audio_is_valid_format(input.config.format)) {
ALOGE("createTrack() invalid format %#x", input.config.format);
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
// 根据 outputId 从 Vector 中调度一个线程 ,没有就返回 default
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;
}
// 创建一个 client ,使用 pid 来 分配内存
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;
// 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,
input.opPackageName);
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
// 将结果包装成 TrackHandle
trackHandle = new TrackHandle(track);
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseOutput(portId);
}
*status = lStatus;
return trackHandle;
}
这个首先组装了一下参数,然后查找了 PlaybackThread 这个应该是播放线程,在 AudioFliger 中发现只有一个openOutput_l中像这个列表中添加了数据,但是没有找到对应的调用位置,不过通过上下文联系这个方法肯定是在createTrack 之前调用的, 然后调用registerPid方法创建client , 使用pid 来管理内存,然后在使用 thread 创建 track,最后将 track 包装成 TrackHandle 返回, 在底层 Handle 是binder 的句柄,那 TrackHandle 是不是把 track 包装成了 binder ,而且这里是根据不同的类型来选择播放的线程,那么就能证明 手机上偶尔存在的同时播放两个声音了的问题了,不同的类型的声音播放在不同的线程上面,
先看一下共享内存是怎么弄的
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::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());
}
接下来继续分析 PlaybackThread::createTrack_l 方法
// 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,
const std::string& opPackageName)
{
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);
}
// Check if requested flags are compatible with output stream flags
if ((*flags & outputFlags) != *flags) {
*flags = (audio_output_flags_t)(*flags & outputFlags);
}
// client expresses a preference for FAST, but we get the final say
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (
// PCM data
audio_is_linear_pcm(format) &&
// TODO: extract as a data library function that checks that a computationally
// expensive downmixer is not required: isFastOutputChannelConversion()
(channelMask == (mChannelMask | mHapticChannelMask) ||
mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
(channelMask == AUDIO_CHANNEL_OUT_MONO
/* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
// hardware sample rate
(sampleRate == mSampleRate) &&
// normal mixer has an associated fast mixer
hasFastMixer() &&
// there are sufficient fast track slots available
(mFastTrackAvailMask != 0)
// FIXME test that MixerThread for this fast track has a capable output HAL
// FIXME add a permission test also?
) {
// static tracks can have any nonzero framecount, streaming tracks check against minimum.
if (sharedBuffer == 0) {
// read the fast track multiplier property the first time it is needed
int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
if (ok != 0) {
ALOGE("%s pthread_once failed: %d", __func__, ok);
}
frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0
}
// check compatibility with audio effects.
{ // scope for mLock
Mutex::Autolock _l(mLock);
for (audio_session_t session : {
AUDIO_SESSION_DEVICE,
AUDIO_SESSION_OUTPUT_STAGE,
AUDIO_SESSION_OUTPUT_MIX,
sessionId,
}) {
sp<EffectChain> chain = getEffectChain_l(session);
if (chain.get() != nullptr) {
audio_output_flags_t old = *flags;
chain->checkOutputFlagCompatibility(flags);
if (old != *flags) {
}
}
}
}
} else {
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
if (!audio_has_proportional_frames(format)) {
if (sharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
frameCount = mNormalFrameCount;
}
if (notificationFrameCount != frameCount) {
notificationFrameCount = frameCount;
}
} else if (sharedBuffer != 0) {
// FIXME: Ensure client side memory buffers need
// not have additional alignment beyond sample
// (e.g. 16 bit stereo accessed as 32 bit frame).
size_t alignment = audio_bytes_per_sample(format);
if (alignment & 1) {
// for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
size_t frameSize = channelCount * audio_bytes_per_sample(format);
if (channelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uintptr_t)sharedBuffer->unsecurePointer() & (alignment - 1)) != 0) {
lStatus = BAD_VALUE;
goto Exit;
}
// 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() / frameSize;
} else {
size_t minFrameCount = 0;
// For fast tracks we try to respect the application's request for notifications per buffer.
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (notificationsPerBuffer > 0) {
// Avoid possible arithmetic overflow during multiplication.
if (notificationsPerBuffer > SIZE_MAX / mFrameCount) {
} else {
minFrameCount = mFrameCount * notificationsPerBuffer;
}
}
} else {
// For normal PCM streaming tracks, update minimum frame count.
// Buffer depth is forced to be at least 2 x the normal mixer frame count and
// cover audio hardware latency.
// This is probably too conservative, but legacy application code may depend on it.
// If you change this calculation, also review the start threshold which is related.
uint32_t latencyMs = latency_l();
if (latencyMs == 0) {
lStatus = UNKNOWN_ERROR;
goto Exit;
}
minFrameCount = AudioSystem::calculateMinFrameCount(latencyMs, mNormalFrameCount,
mSampleRate, sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
}
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
}
// Make sure that application is notified with sufficient margin before underrun.
// The client can divide the AudioTrack buffer into sub-buffers,
// and expresses its desire to server as the notification frame count.
if (sharedBuffer == 0 && audio_is_linear_pcm(format)) {
size_t maxNotificationFrames;
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
// notify every HAL buffer, regardless of the size of the track buffer
maxNotificationFrames = mFrameCount;
} else {
// Triple buffer the notification period for a triple buffered mixer period;
// otherwise, double buffering for the notification period is fine.
//
// TODO: This should be moved to AudioTrack to modify the notification period
// on AudioTrack::setBufferSizeInFrames() changes.
const int nBuffering =
(uint64_t{frameCount} * mSampleRate)
/ (uint64_t{mNormalFrameCount} * sampleRate) == 3 ? 3 : 2;
maxNotificationFrames = frameCount / nBuffering;
// If client requested a fast track but this was denied, then use the smaller maximum.
if (requestedFlags & AUDIO_OUTPUT_FLAG_FAST) {
size_t maxNotificationFramesFastDenied = FMS_20 * sampleRate / 1000;
if (maxNotificationFrames > maxNotificationFramesFastDenied) {
maxNotificationFrames = maxNotificationFramesFastDenied;
}
}
}
if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) {
if (notificationFrameCount == 0) {
} else {
}
notificationFrameCount = maxNotificationFrames;
}
}
*pFrameCount = frameCount;
*pNotificationFrameCount = notificationFrameCount;
switch (mType) {
case DIRECT:
if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
lStatus = BAD_VALUE;
goto Exit;
}
}
break;
case OFFLOAD:
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
lStatus = BAD_VALUE;
goto Exit;
}
break;
default:
if (!audio_is_linear_pcm(format)) {
lStatus = BAD_VALUE;
goto Exit;
}
if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
lStatus = BAD_VALUE;
goto Exit;
}
break;
}
lStatus = initCheck();
if (lStatus != NO_ERROR) {
goto Exit;
}
{ // scope for mLock
Mutex::Autolock _l(mLock);
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0 && t->isExternalTrack()) {
uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
if (sessionId == t->sessionId() && strategy != actual) {
lStatus = BAD_VALUE;
goto Exit;
}
}
}
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,
SIZE_MAX /*frameCountToBeReady*/, opPackageName);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
// 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(track, callback);
}
}
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
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;
}
到了这里就是将参数交给 创建后的track,
再开看一下 TrackHandle 先来看一下他的类的结构
class TrackHandle : public android::BnAudioTrack {
public:
explicit TrackHandle(const sp<PlaybackThread::Track>& track);
virtual ~TrackHandle();
virtual sp<IMemory> getCblk() const;
virtual status_t start();
virtual void stop();
virtual void flush();
virtual void pause();
virtual status_t attachAuxEffect(int effectId);
virtual status_t setParameters(const String8& keyValuePairs);
virtual status_t selectPresentation(int presentationId, int programId);
virtual media::VolumeShaper::Status applyVolumeShaper(
const sp<media::VolumeShaper::Configuration>& configuration,
const sp<media::VolumeShaper::Operation>& operation) override;
virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) override;
virtual status_t getTimestamp(AudioTimestamp& timestamp);
virtual void signal(); // signal playback thread for a change in control block
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
const sp<PlaybackThread::Track> mTrack;
};
再看看 BnAudioTrack 这个类
class BnAudioTrack 这个类 : public BnInterface<IAudioTrack>
{
public:
virtual status_t onTransact( uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags = 0);
};
到了这个初始化的工作已经完成了,
