最近在看Android中播放延迟的问题,看了下代码,发现AudioTrack类中的函数latency有以下注释:
/* Returns this track's latency in milliseconds. * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) * and audio hardware driver. */
uint32_t AudioTrack::latency() const { return mLatency; }
mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
uint32_t afLatency; if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { return NO_INIT; }
status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType) { OutputDescriptor *outputDesc; audio_io_handle_t output; if (streamType == DEFAULT) { streamType = MUSIC; } output = getOutput((stream_type)streamType); if (output == 0) { return PERMISSION_DENIED; } gLock.lock(); outputDesc = AudioSystem::gOutputs.valueFor(output); if (outputDesc == 0) { gLock.unlock(); const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; *latency = af->latency(output); } else { *latency = outputDesc->latency; gLock.unlock(); } LOGV("getOutputLatency() streamType %d, output %d, latency %d", streamType, output, *latency); return NO_ERROR; }
return thread->latency();
uint32_t AudioFlinger::PlaybackThread::latency() const { if (mOutput) { return mOutput->latency(); } else { return 0; } }
#define USEC_TO_MSEC(x) ((x + 999) / 1000) uint32_t AudioStreamOutALSA::latency() const { // 将微秒转化为毫秒 // Android wants latency in milliseconds. return USEC_TO_MSEC (mHandle->latency); }
out = new AudioStreamOutALSA(this, &(*it));
ALSAHandleList::iterator it = mDeviceList.begin();
mALSADevice->init(mALSADevice, mDeviceList);
static status_t s_init(alsa_device_t *module, ALSAHandleList &list) { LOGD("Initializing devices for IMX51 ALSA module"); list.clear(); for (size_t i = 0; i < ARRAY_SIZE(_defaults); i++) { _defaults[i].module = module; list.push_back(_defaults[i]); } return NO_ERROR; }
_defaults的定义:
static alsa_handle_t _defaults[] = { { module : 0, devices : IMX51_OUT_DEFAULT, curDev : 0, curMode : 0, handle : 0, format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT channels : 2, sampleRate : DEFAULT_SAMPLE_RATE, latency : 200000, // Desired Delay in usec bufferSize : 6144, // Desired Number of samples modPrivate : (void *)&setDefaultControls, }, { module : 0, devices : IMX51_IN_DEFAULT, curDev : 0, curMode : 0, handle : 0, format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT channels : 2, sampleRate : DEFAULT_SAMPLE_RATE, latency : 250000, // Desired Delay in usec bufferSize : 6144, // Desired Number of samples modPrivate : (void *)&setDefaultControls, }, };
latency : 200000, // Desired Delay in usec
mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice, &outputDesc->mSamplingRate, &outputDesc->mFormat, &outputDesc->mChannels, &outputDesc->mLatency, outputDesc->mFlags);
if (pLatencyMs) *pLatencyMs = thread->latency();