Android杂谈:AudioFlinger与AudioTrac
2017-05-28 本文已影响302人
我在等你回复可你没回
写这个玩意纯粹是开个头啦
AT是客户端，AF是服务端。
/frameworks/av/media/libmedia/AudioTrack.cpp
AudioTrack构造函数，初始化一些变量，执行set函数。好多变量都不知道是啥。
AudioTrack::AudioTrack(
audio_stream_type_t streamType, //streamType，例如music
uint32_t sampleRate, //采样率
audio_format_t format, //格式
audio_channel_mask_t channelMask, //频道掩码，什么鬼
size_t frameCount, //每个buf的帧数
audio_output_flags_t flags, //什么鬼
callback_t cbf, //回调
void* user, //不知道
uint32_t notificationFrames, //不知道
int sessionId, //会话
transfer_type transferType, //不知道
const audio_offload_info_t *offloadInfo, //不知道
int uid,
pid_t pid,
const audio_attributes_t* pAttributes, //属性
bool doNotReconnect)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), //优先级
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
offloadInfo, uid, pid, pAttributes, doNotReconnect);
}
看看漫长的set函数
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,
uint32_t notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect)
{
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %u, sessionId %d, transferType %d, uid %d, pid %d",
streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
sessionId, transferType, uid, pid);
switch (transferType) {
//音频传送方式，播放音乐是TRANSFER_DEFAULT，播放fm是TRANSFER_SYNC
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:
if (cbf == NULL || sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
return BAD_VALUE;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
return BAD_VALUE;
}
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mSharedBuffer = sharedBuffer;//buf大小
mTransfer = transferType;
mDoNotReconnect = doNotReconnect;
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(),
sharedBuffer->size());
ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags);
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) { //避免重复创建
ALOGE("Track already in use");
return INVALID_OPERATION;
}
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (pAttributes == NULL) {
if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) { //检验streamType有效性
ALOGE("Invalid stream type %d", streamType);
return BAD_VALUE;
}
mStreamType = streamType;
} else {
// stream type shouldn't be looked at, this track has audio attributes //audio属性不知道是什么鬼
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
mStreamType = AUDIO_STREAM_DEFAULT;
if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
}
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %#x", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
// force direct flag if format is not linear PCM
// or offload was requested
//direct flag是什么鬼
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
|| !audio_is_linear_pcm(format)) {
ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
? "Offload request, forcing to Direct Output"
: "Not linear PCM, forcing to Direct Output");
flags = (audio_output_flags_t)
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// force direct flag if HW A/V sync requested 硬件同步
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);//frame大小
} else {
mFrameSize = sizeof(uint8_t);
}
} else {
ALOG_ASSERT(audio_is_linear_pcm(format));
mFrameSize = channelCount * audio_bytes_per_sample(format);
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
}
// sampling rate must be specified for direct outputs
//采样率，直连输出模式必须指定采样率
if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
return BAD_VALUE;
}
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
//offload的东东
// 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;
}
//左右声道音量
mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
mSendLevel = 0.0f;
// mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
//sessionId 赋值
if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = AudioSystem::newAudioUniqueId();
} else {
mSessionId = sessionId;
}
//pid uid
int callingpid = IPCThreadState::self()->getCallingPid();
int mypid = getpid();
if (uid == -1 || (callingpid != mypid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingpid != mypid)) {
mClientPid = callingpid;
} else {
mClientPid = pid;
}
mAuxEffectId = 0;
mFlags = flags;
mCbf = cbf;
//创建AudioTrackThread线程
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioTrack
//创建IAudioTrack
status_t status = createTrack_l();
//出现错误时退出AudioTrackThread线程
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
return status;
}
//变量赋值
mStatus = NO_ERROR;
mState = STATE_STOPPED;
mUserData = user;
mLoopCount = 0;
mLoopStart = 0;
mLoopEnd = 0;
mLoopCountNotified = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mPosition = 0;
mReleased = 0;
mStartUs = 0;
AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
mPreviousTimestampValid = false;
mTimestampStartupGlitchReported = false;
mRetrogradeMotionReported = false;
return NO_ERROR;
}
createTrack_l函数会调用AudioFlinger创建IAudioTrack
status_t AudioTrack::createTrack_l()
{
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); //获取IAudioFlinger
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
return NO_INIT;
}
if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 清理mDeviceCallback mOutput
AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput);
}
audio_io_handle_t output;
audio_stream_type_t streamType = mStreamType;
audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL;
status_t status;
//获取output
status = AudioSystem::getOutputForAttr(attr, &output,
(audio_session_t)mSessionId, &streamType, mClientUid,
mSampleRate, mFormat, mChannelMask,
mFlags, mSelectedDeviceId, mOffloadInfo);
//获取output失败
if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x,"
" channel mask %#x, flags %#x",
mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
return BAD_VALUE;
}
{
// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
// we must release it ourselves if anything goes wrong.
// Not all of these values are needed under all conditions, but it is easier to get them all
status = AudioSystem::getLatency(output, &mAfLatency);
if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
goto release;
}
//Latency是什么鬼
ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency);
//帧数
status = AudioSystem::getFrameCount(output, &mAfFrameCount);
if (status != NO_ERROR) {
ALOGE("getFrameCount(output=%d) status %d", output, status);
goto release;
}
//采样率
status = AudioSystem::getSamplingRate(output, &mAfSampleRate);
if (status != NO_ERROR) {
ALOGE("getSamplingRate(output=%d) status %d", output, status);
goto release;
}
if (mSampleRate == 0) {
mSampleRate = mAfSampleRate;
mOriginalSampleRate = mAfSampleRate;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !((
// either of these use cases:
// use case 1: shared buffer
(mSharedBuffer != 0) ||
// use case 2: callback transfer mode
(mTransfer == TRANSFER_CALLBACK) ||
// use case 3: obtain/release mode
(mTransfer == TRANSFER_OBTAIN)) &&
// matching sample rate
(mSampleRate == mAfSampleRate))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, output %u Hz",
mTransfer, mSampleRate, mAfSampleRate);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
// n = 1 fast track with single buffering; nBuffering is ignored
// n = 2 fast track with double buffering
// n = 2 normal track, (including those with sample rate conversion)
// n >= 3 very high latency or very small notification interval (unused).
const uint32_t nBuffering = 2;
mNotificationFramesAct = mNotificationFramesReq;
//帧数
size_t frameCount = mReqFrameCount;
if (!audio_is_linear_pcm(mFormat)) {
//buf大小
if (mSharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = mSharedBuffer->size();
} else if (frameCount == 0) {
frameCount = mAfFrameCount;
}
if (mNotificationFramesAct != frameCount) {
mNotificationFramesAct = frameCount;
}
} else if (mSharedBuffer != 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(mFormat);
if (alignment & 1) {
// for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
if (mChannelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
mSharedBuffer->pointer(), mChannelCount);
status = BAD_VALUE;
goto release;
}
// 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 = mSharedBuffer->size() / mFrameSize;
} else {
// For fast tracks the frame count calculations and checks are done by server
if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
// for normal tracks precompute the frame count based on speed.
const size_t minFrameCount = calculateMinFrameCount(
mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
mPlaybackRate.mSpeed);
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
}
}
//trace标志
IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
if (mIsTimed) {
trackFlags |= IAudioFlinger::TRACK_TIMED;
}
pid_t tid = -1;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
trackFlags |= IAudioFlinger::TRACK_DIRECT;
}
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
// but we will still need the original value also
int originalSessionId = mSessionId;
//关键，利用audioFlinger创建IAudioTrack
sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
mSampleRate,
mFormat,
mChannelMask,
&temp,
&trackFlags,
mSharedBuffer,
output,
tid,
&mSessionId,
mClientUid,
&status);
ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
"session ID changed from %d to %d", originalSessionId, mSessionId);
if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create track, status: %d", status);
goto release;
}
ALOG_ASSERT(track != 0);
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
//共享内存
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
return NO_INIT;
}
void *iMemPointer = iMem->pointer();
if (iMemPointer == NULL) {
ALOGE("Could not get control block pointer");
return NO_INIT;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
mAudioTrack = track;
mCblkMemory = iMem;
IPCThreadState::self()->flushCommands();
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
// note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
// In current design, AudioTrack client checks and ensures frame count validity before
// passing it to AudioFlinger so AudioFlinger should not return a different value except
// for fast track as it uses a special method of assigning frame count.
ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
}
frameCount = temp;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
// FIXME This is a warning, not an error, so don't return error status
//return NO_INIT;
}
}
if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (trackFlags & IAudioFlinger::TRACK_DIRECT) {
ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server");
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT);
// FIXME This is a warning, not an error, so don't return error status
//return NO_INIT;
}
}
// Make sure that application is notified with sufficient margin before underrun
if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
// Theoretically double-buffering is not required for fast tracks,
// due to tighter scheduling. But in practice, to accommodate kernels with
// scheduling jitter, and apps with computation jitter, we use double-buffering
// for fast tracks just like normal streaming tracks.
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount / nBuffering) {
mNotificationFramesAct = frameCount / nBuffering;
}
}
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output;
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 {
buffers = mSharedBuffer->pointer();
if (buffers == NULL) {
ALOGE("Could not get buffer pointer");
return NO_INIT;
}
}
//音效
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME doesn't take into account speed or future sample rate changes (until restoreTrack)
// FIXME don't believe this lie
mLatency = mAfLatency + (1000*frameCount) / mSampleRate;
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (frameCount > mReqFrameCount) {
mReqFrameCount = frameCount;
}
// reset server position to 0 as we have new cblk.
mServer = 0;
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
mProxy = mStaticProxy;
}
//代理
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);
if (mDeviceCallback != 0) {
AudioSystem::addAudioDeviceCallback(mDeviceCallback, mOutput);
}
return NO_ERROR;
}
release:
AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId);
if (status == NO_ERROR) {
status = NO_INIT;
}
return status;
}
/frameworks/av/services/audioflinger/AudioFlinger.cpp
AudioFlinger构造函数，初始化一些变量，感觉没做什么事
AudioFlinger::AudioFlinger()
: BnAudioFlinger(), //继承与BnAudioFlinger
mPrimaryHardwareDev(NULL), //主设备
mAudioHwDevs(NULL), //音频硬件设备
mHardwareStatus(AUDIO_HW_IDLE), //设备状态，初始化为空闲
mMasterVolume(1.0f), //主音量
mMasterMute(false), //静音
mNextUniqueId(1),
mMode(AUDIO_MODE_INVALID), //mode
mBtNrecIsOff(false),
mIsLowRamDevice(true), //低内存设备
mIsDeviceTypeKnown(false),
mGlobalEffectEnableTime(0),
mSystemReady(false)
{ //构造函数没做什么
getpid_cached = getpid();
char value[PROPERTY_VALUE_MAX];
bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
if (doLog) {
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
MemoryHeapBase::READ_ONLY);
}
#ifdef TEE_SINK
(void) property_get("ro.debuggable", value, "0");
int debuggable = atoi(value);
int teeEnabled = 0;
if (debuggable) {
(void) property_get("af.tee", value, "0");
teeEnabled = atoi(value);
}
// FIXME symbolic constants here
if (teeEnabled & 1) {
mTeeSinkInputEnabled = true;
}
if (teeEnabled & 2) {
mTeeSinkOutputEnabled = true;
}
if (teeEnabled & 4) {
mTeeSinkTrackEnabled = true;
}
#endif
}
AudioFlinger初始化，设置了mode为AUDIO_MODE_NORMAL
void AudioFlinger::onFirstRef()
{
int rc = 0;
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
char val_str[PROPERTY_VALUE_MAX] = { 0 };
if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
uint32_t int_val;
if (1 == sscanf(val_str, "%u", &int_val)) {
mStandbyTimeInNsecs = milliseconds(int_val);
ALOGI("Using %u mSec as standby time.", int_val);
} else {
mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
ALOGI("Using default %u mSec as standby time.",
(uint32_t)(mStandbyTimeInNsecs / 1000000));
}
}
mPatchPanel = new PatchPanel(this);
mMode = AUDIO_MODE_NORMAL;
}
AudioFlinger代码结构图
AudioFlinger.png
DEMO
传送门：https://github.com/wenfengtou/AudioDemo
一个简单的使用audiorecord与audiotrace的demo
demo
录音开始到结束的log
12-31 20:14:32.669 14860 17009 D AudioRecord: getMinFrameCount 1024
12-31 20:14:32.670 14860 17009 D AudioRecord: set(): inputSource 1, sampleRate 8000, format 0x1, channelMask 0x10, frameCount 1024, notificationFrames 0, sessionId 0, transferType 0, flags 0, opPackageName com.wenfengtou.audiodemo uid -1, pid -1
12-31 20:14:32.675 14860 17009 D AudioRecord: set: Create AudioRecordThread
12-31 20:14:32.677 14860 17009 D AudioRecord: openRecord_l
12-31 20:14:32.678 513 1348 V APM_AudioPolicyManager: getInputForAttr() source 1, samplingRate 8000, format 1, channelMask 10,session 41, flags 0
12-31 20:14:32.680 513 1348 D AudioALSAStreamManager: +openInputStream()
12-31 20:14:32.680 513 1348 D AudioALSAStreamManager: openInputStream(), devices = 0x80000004, format = 0x1, channels = 0x10, sampleRate = 8000, status = 38, acoustics = 0x0
12-31 20:14:32.680 513 1348 D AudioALSAStreamIn: AudioALSAStreamIn()
12-31 20:14:32.680 513 1348 D AudioALSAStreamIn: set(), devices = 0x80000004, format = 0x1, channels = 0x10, sampleRate = 8000, acoustics = 0x0, flags = 0
12-31 20:14:32.680 513 1348 D AudioALSAStreamIn: +CheckBesRecordInfo()
12-31 20:14:32.680 513 1348 D AudioSpeechEnhanceInfo: GetBesRecScene() -1
12-31 20:14:32.680 513 1348 D AudioSpeechEnhanceInfo: ResetBesRecScene()
12-31 20:14:32.680 513 1348 D AudioSpeechEnhanceInfo: IsBesRecTuningEnable()+
12-31 20:14:32.681 513 1348 D AudioSpeechEnhanceInfo: IsBesRecTuningEnable()- 0
12-31 20:14:32.681 513 1348 D AudioYusuParam: QueryFeatureSupportInfo(),feature support a090055
12-31 20:14:32.681 513 1348 D AudioSpeechEnhanceInfo: GetBesRecVMFileName()+
12-31 20:14:32.681 513 1348 D AudioSpeechEnhanceInfo: GetBesRecVMFileName(), mVMFileName=, VMFileName=
12-31 20:14:32.681 513 1348 D AudioYusuParam: QueryFeatureSupportInfo(),feature support a090055
12-31 20:14:32.681 513 1348 D AudioSpeechEnhanceInfo: GetForceMagiASRState(), feature_support=a090055, 0, mForceMagiASR=0
12-31 20:14:32.681 513 1348 D AudioSpeechEnhanceInfo: GetForceAECRecState(), mForceAECRec=0
12-31 20:14:32.681 513 1348 D AudioALSAStreamIn: -CheckBesRecordInfo()
12-31 20:14:32.681 513 1348 D AudioALSAStreamIn: checkOpenStreamSampleRate(), origin sampleRate 8000, kDefaultInputSourceSampleRate 48000.
12-31 20:14:32.681 513 1348 D AudioALSAStreamIn: set() done, sampleRate = 8000, num_channels = 1, buffer_size=320
12-31 20:14:32.681 513 1348 E AudioALSAStreamManager: openInputStream(), SetInputMute(mAllInputMute)
12-31 20:14:32.683 513 1348 D AudioALSAStreamIn: SetInputMute(), 0
12-31 20:14:32.683 513 1348 D AudioALSAStreamIn: -SetInputMute()
12-31 20:14:32.685 513 1348 D AudioALSAStreamManager: -openInputStream(), in = 0xe7114180, status = 0x0, mStreamInVector.size() = 2
12-31 20:14:32.685 513 1348 D AudioALSAStreamIn: +setParameters(): input_source=1
12-31 20:14:32.686 513 1348 D AudioALSAStreamIn: setParameters() InputSource = 1
12-31 20:14:32.686 513 1348 D AudioALSAStreamIn: -setParameters(): input_source=1
12-31 20:14:32.699 513 17011 I AudioFlinger_Threads: AudioFlinger's thread 0xe6283700 ready to run
12-31 20:14:32.700 513 17011 D AudioALSAStreamIn: +standby()
12-31 20:14:32.701 513 17011 D AudioALSAStreamIn: -standby()
12-31 20:14:32.711 513 17011 D AudioALSAStreamIn: +standby()
12-31 20:14:32.711 513 17011 D AudioALSAStreamIn: -standby()
12-31 20:14:32.720 513 1348 V AudioPolicyService: AudioCommandThread() adding update audio port list
12-31 20:14:32.720 513 1348 V APM_AudioPolicyManager: getInputForAttr() returns input type = 0 device =0x80000004 *input = 38
12-31 20:14:32.721 513 794 V AudioPolicyService: AudioCommandThread() processing update audio port list
12-31 20:14:32.741 513 1349 D AudioFlinger: track(0xe6f83180): mIsOut 0, mFrameCount 1024, mSampleRate 8000, mFormat 1, mChannelCount 1, mTrackCount 5, thread 0xe6283700, sessionId 41
12-31 20:14:32.745 14860 17009 D AudioRecord: start, sync event 0 trigger session 0
12-31 20:14:32.763 513 513 V APM_AudioPolicyManager: startInput() input 38
12-31 20:14:32.763 513 793 V AudioPolicyService: AudioCommandThread() processing create audio patch
12-31 20:14:32.766 513 17011 D AudioALSAHardware: +createAudioPatch num_sources [1] , num_sinks [1]
12-31 20:14:32.766 513 17011 D AudioALSAHardware: +routing createAudioPatch 80000004->Mixer Src 1
12-31 20:14:32.766 513 17011 D AudioALSAStreamManager: +setParameters(), IOport = 38, keyValuePairs = input_source=1;routing=-2147483644
12-31 20:14:32.766 513 17011 D AudioALSAStreamManager: Send to mStreamInVector [1]
12-31 20:14:32.766 513 17011 D AudioALSAStreamIn: +setParameters(): input_source=1;routing=-2147483644
12-31 20:14:32.767 513 17011 D AudioALSAStreamIn: setParameters() InputSource = 1
12-31 20:14:32.767 513 17011 D AudioALSAStreamManager: +routingInputDevice(), input_device: 0x80000004 => 0x80000004
12-31 20:14:32.767 513 17011 W AudioALSAStreamManager: -routingInputDevice(), input_device == current_input_device(0x80000004), return
12-31 20:14:32.767 513 17011 D AudioALSAStreamIn: -setParameters(): input_source=1;routing=-2147483644
12-31 20:14:32.767 513 17011 D AudioALSAStreamManager: -setParameters()
12-31 20:14:32.767 513 17011 D AudioALSAHardware: handlecheck createAudioPatch sucess new *handle 0x4
12-31 20:14:32.773 513 513 V APM_AudioPolicyManager: setInputDevice() createAudioPatch returned 0 patchHandle 36
12-31 20:14:32.774 513 513 V APM_AudioPolicyManager: AudioPolicyManager::startInput() input source = 1
12-31 20:14:32.774 513 513 V AudioPolicyService: AudioCommandThread() adding recording configuration update event 1, source 1
12-31 20:14:32.775 513 794 V AudioPolicyService: AudioCommandThread() processing recording configuration update
12-31 20:14:32.777 513 17011 D AudioALSAStreamIn: open()
12-31 20:14:32.777 513 17011 D AudioALSAStreamManager: +createCaptureHandler(), mAudioMode = 0, input_source = 1, input_device = 0x80000004, mBypassDualMICProcessUL=0, sample_rate=8000
12-31 20:14:32.777 513 17011 D AudioYusuParam: QueryFeatureSupportInfo(),feature support a090055
12-31 20:14:32.777 513 17011 D AudioALSAStreamManager: -EnableBesRecord(), 1
12-31 20:14:32.778 513 17011 D AudioALSACaptureHandlerBase: AudioALSACaptureHandlerBase()
12-31 20:14:32.778 513 17011 D AudioALSACaptureHandlerBase: init()
12-31 20:14:32.778 513 17011 D AudioALSACaptureHandlerNormal: AudioALSACaptureHandlerNormal()
12-31 20:14:32.778 513 17011 D AudioALSACaptureHandlerNormal: init()
12-31 20:14:32.778 513 17011 D AudioALSAStreamManager: -createCaptureHandler(), mCaptureHandlerVector.size() = 1
12-31 20:14:32.778 513 17011 D AudioALSACaptureHandlerNormal: +open(), input_device = 0x80000004, input_source = 0x1, sample_rate=8000, num_channels=1
12-31 20:14:32.778 513 17011 D AudioALSACaptureDataClient: AudioALSACaptureDataClient()
12-31 20:14:32.778 513 17011 D AudioSPELayer: SPELayer+
12-31 20:14:32.778 513 17011 D AudioSPELayer: GetVoIPJitterTime JitterTime=0,ret=1
12-31 20:14:32.783 513 17011 D AudioSPELayer: SPELayer-
12-31 20:14:32.784 513 17011 D AudioMTKGainController: +SetCaptureGain(), mode=0, source=1, input device=0x80000004, output device=0x2
12-31 20:14:32.784 513 17011 D AudioYusuParam: QueryFeatureSupportInfo(),feature support a090055
12-31 20:14:32.784 513 17011 D AudioMTKGainController: getGainDevice(), input devices = 0x80000004, return gainDevice = 2
12-31 20:14:32.784 513 17011 D AudioMTKGainController: ApplyMicGain(), _mic_mode = 0, _gain_device = 2, mode = 0, micgain = 26, mULTotalGain = 176
12-31 20:14:32.784 513 17011 D AudioMTKGainController: SetAdcPga1 = 3
12-31 20:14:32.784 513 17011 D AudioMTKGainController: SetAdcPga2 = 3
12-31 20:14:32.785 513 17011 D AudioALSACaptureDataClient: AudioALSACaptureDataClient(), besrecord_enable=1, besrecord_scene=-1
12-31 20:14:32.785 513 17011 D AudioSPELayer: SetVMDumpEnable(), 0
12-31 20:14:32.785 513 17011 D AudioSPELayer: SetVMDumpFileName()+, VMFileName=
12-31 20:14:32.783 14860 17009 D AudioRecord: return status 0
12-31 20:14:32.785 513 17011 D AudioSPELayer: SetVMDumpFileName()-, VMFileName=, mVMDumpFileName=
12-31 20:14:32.785 513 17011 D AudioALSACaptureDataClient: CheckBesRecordBypass() 0
12-31 20:14:32.785 513 17011 D AudioALSACaptureDataClient: +ConfigBesRecordParams()
12-31 20:14:32.785 513 17011 D AudioALSACaptureDataClient: +GetBesRecordRoutePath(), output device = 0x2, input device = 0x80000004
12-31 20:14:32.785 513 17011 D AudioYusuParam: QueryFeatureSupportInfo(),feature support a090055
12-31 20:14:32.785 513 17011 D AudioALSACaptureDataClient: CheckBesRecordStereoModeEnable(), mBesRecordStereoMode = 0
12-31 20:14:32.785 513 17011 D AudioALSACaptureDataClient: ConfigBesRecordParams(), categoryPath = Application,SndRecNormal,Profile,Handset, mBesRecordStereoMode = 0, input_source = 1, input_devices = 80000004, bVoIPEnable = 0, bypassDualProcess = 0
12-31 20:14:32.785 513 17011 D AudioSPELayer: SetEnhPara, SPE_MODE=1
12-31 20:14:32.786 513 17011 D AudioSPELayer: SetCompFilter, SPE_MODE=1
12-31 20:14:32.786 513 17011 D AudioYusuParam: QueryFeatureSupportInfo(),feature support a090055
12-31 20:14:32.786 513 17011 D AudioSPELayer: SetDMNRPara, SPE_MODE=1
12-31 20:14:32.786 513 17011 D AudioALSACaptureDataClient: SetDMNREnable(), type=0, bypassDMNR=0
12-31 20:14:32.786 513 17011 D AudioYusuParam: QueryFeatureSupportInfo(),feature support a090055
12-31 20:14:32.786 513 17011 D AudioSPELayer: SetSampleRate, SPE_MODE=1
12-31 20:14:32.786 513 17011 D AudioSPELayer: SetAPPTable, SPE_MODE=1, App_table=4
12-31 20:14:32.786 513 17011 D AudioSPELayer: SetRoute, 2
12-31 20:14:32.786 513 17011 D AudioSPELayer: SetMICDigitalGain, MIC_DG, SPE_MODE=1, gain=12
12-31 20:14:32.786 513 17011 D AudioSPELayer: SetUpLinkTotalGain, SPE_MODE=1, gain=176
12-31 20:14:32.786 513 17011 D AudioALSACaptureDataClient: -ConfigBesRecordParams()
12-31 20:14:32.786 513 17011 D AudioALSACaptureDataClient: +StartBesRecord()
12-31 20:14:32.786 513 17011 D AudioSPELayer: Start+, mode=1
12-31 20:14:32.787 513 17011 D AudioSPELayer: mRecordSampleRate=48000, mRecordFrameRate=20, mRecordApp_table=4
12-31 20:14:32.787 513 17011 D AudioSPELayer: Start, SWIP ver = 9.1
12-31 20:14:32.787 513 17011 D AudioSPELayer: Start, going to configure,mSphCtrlBuffer=0xe6349d80,mem_size=121536
12-31 20:14:32.787 513 17011 D AudioSPELayer: dump, State=0, mode=0
12-31 20:14:32.787 513 17011 D AudioSPELayer: Record:Samplerate = 48000, FrameRate=20,App_table=4, Fea_Cfg_table=1ff
12-31 20:14:32.787 513 17011 D AudioSPELayer: Record:EnhanceParas
12-31 20:14:32.787 513 17011 D AudioSPELayer: 0
12-31 20:14:32.787 513 17011 D AudioSPELayer: 479
12-31 20:14:32.788 513 17011 D AudioSPELayer: 16388
12-31 20:14:32.788 513 17011 D AudioSPELayer: 36892
12-31 20:14:32.788 513 17011 D AudioSPELayer: 37639
12-31 20:14:32.788 513 17011 D AudioSPELayer: 8192
12-31 20:14:32.788 513 17011 D AudioSPELayer: 896
12-31 20:14:32.788 513 17011 D AudioSPELayer: 4
12-31 20:14:32.788 513 17011 D AudioSPELayer: 36624
12-31 20:14:32.788 513 17011 D AudioSPELayer: 197
12-31 20:14:32.788 513 17011 D AudioSPELayer: 611
12-31 20:14:32.788 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 8192
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.789 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.790 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.791 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 55997
12-31 20:14:32.792 513 17011 D AudioSPELayer: 31000
12-31 20:14:32.792 513 17011 D AudioSPELayer: 10752
12-31 20:14:32.792 513 17011 D AudioSPELayer: 32769
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.792 513 17011 D AudioSPELayer: 0
12-31 20:14:32.793 513 17011 D AudioSPELayer: 0
12-31 20:14:32.793 513 17011 D AudioSPELayer: 0
12-31 20:14:32.793 513 17011 D AudioSPELayer: 0
12-31 20:14:32.793 513 17011 D AudioSPELayer: 0
12-31 20:14:32.793 513 17011 D AudioSPELayer: 0
12-31 20:14:32.793 513 17011 D AudioSPELayer: Using:Samplerate = 48000, FrameRate=20,MIC_DG=12, App_table=4, Fea_Cfg_table=1ff, MMI_table=ffffffff, Device_mode=2, MMI_MIC_GAIN=176
12-31 20:14:32.793 513 17011 D AudioSPELayer: Using:EnhanceParas
12-31 20:14:32.793 513 17011 D AudioSPELayer: [index 0] 0,479,16388,36892,37639,8192,896
12-31 20:14:32.793 513 17011 D AudioSPELayer: [index 1] 4,36624,197,611,0,0,0
12-31 20:14:32.793 513 17011 D AudioSPELayer: [index 2] 0,8192,0,0,0,0,0
12-31 20:14:32.793 513 17011 D AudioSPELayer: [index 3] 0,0,0,0,0,0,0
12-31 20:14:32.794 513 17011 D AudioSPELayer: [index 4] 0,0,0,0,0,0,0
12-31 20:14:32.794 513 17011 D AudioSPELayer: [index 5] 0,0,0,0,0,0,0
12-31 20:14:32.794 513 17011 D AudioSPELayer: [index 6] 0,0,0,0,0,0,0
12-31 20:14:32.794 513 17011 D AudioSPELayer: [index 7] 55997,31000,10752,32769,0,0,0
12-31 20:14:32.794 513 17011 D AudioSPELayer: Using:DMNRCalData
12-31 20:14:32.794 513 17011 D AudioSPELayer: [index 0] 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
12-31 20:14:32.794 513 17011 D AudioSPELayer: [index 1] 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
12-31 20:14:32.794 513 17011 D AudioSPELayer: [index 2] 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
12-31 20:14:32.795 513 17011 D AudioSPELayer: [index 3] 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
12-31 20:14:32.795 513 17011 D AudioSPELayer: mSph_Enh_ctrl.frame_rate 20
12-31 20:14:32.795 513 17011 D AudioSPELayer: mNsecPerSample=62500
12-31 20:14:32.795 513 17011 D AudioSPELayer: mSPEProcessBufSize=3840
12-31 20:14:32.796 513 17011 D AudioALSACaptureDataClient: -StartBesRecord()
12-31 20:14:32.796 513 17011 D AudioALSACaptureDataClient: sample rate = 8000, drop ms = 25, channels = 1, byts per sample = 2, dropBesRecordDataSize = 400
12-31 20:14:32.796 513 17011 D AudioPreProcess: AudioPreProcess()+
12-31 20:14:32.796 513 17011 D AudioPreProcess: AudioPreProcess()-
12-31 20:14:32.796 513 17011 D AudioALSACaptureDataProviderBase: attach(), 0xe6699000
12-31 20:14:32.796 513 17011 D AudioALSACaptureDataClient: setIdentity(), mIsIdentitySet=0, identity=1
12-31 20:14:32.796 513 17011 D AudioALSACaptureDataProviderBase: attach(), mCaptureDataClientIndex=1, mCaptureDataClientVector.size()=0, Identity=1
12-31 20:14:32.797 513 17011 D AudioALSACaptureDataProviderNormal: open()
12-31 20:14:32.797 513 17011 D AudioALSADeviceParser: GetPcmIndexByString() stringpair = MultiMedia1_Capture
12-31 20:14:32.797 513 17011 D AudioALSADeviceParser: compare success = 1
12-31 20:14:32.797 513 17011 D AudioALSADeviceParser: GetCardIndexByString() stringpair = MultiMedia1_Capture
12-31 20:14:32.797 513 17011 D AudioALSADeviceParser: compare success Cardindex = 0
12-31 20:14:32.797 513 17011 D AudioALSACaptureDataProviderNormal: open cardindex = 0 pcmindex = 1
12-31 20:14:32.801 513 17011 D AudioALSACaptureDataProviderNormal: buffersizemax = 49152
12-31 20:14:32.801 513 17011 D AudioSpeechEnhanceInfo: GetHifiRecord, mHiFiRecordEnable=0
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataProviderNormal: bHifiRecord = 0
12-31 20:14:32.802 513 17011 D AudioSpeechEnhanceInfo: GetDebugStatus, mDebugflag=0
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataProviderNormal: btempDebug = 0
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataProviderNormal: getLatencyTime(), audiomode=0
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataProviderBase: +HasLowLatencyCapture()
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataClient: +IsLowLatencyCapture()
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataClient: -IsLowLatencyCapture(), bRet=0
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataProviderBase: -HasLowLatencyCapture(), bRet=0
12-31 20:14:32.802 513 17011 D AudioALSACaptureDataProviderNormal: open(), format = 1, channels=2, rate=48000, period_size=960, period_count=4,latency=20 kReadBufferSize=7680
12-31 20:14:32.803 513 17011 D AudioALSACaptureDataProviderNormal: open(), mCaptureDropSize=0, CAPTURE_DROP_MS=0
12-31 20:14:32.803 513 17011 D AudioALSACaptureDataProviderNormal: open(), period_count=4, period_size=960, samplerate = 48000
12-31 20:14:32.803 513 17011 D AudioALSADeviceParser: GetPcmIndexByString() stringpair = MultiMedia1_Capture
12-31 20:14:32.803 513 17011 D AudioALSADeviceParser: compare success = 1
12-31 20:14:32.803 513 17011 D AudioALSADeviceParser: GetCardIndexByString() stringpair = MultiMedia1_Capture
12-31 20:14:32.803 513 17011 D AudioALSADeviceParser: compare success Cardindex = 0
12-31 20:14:32.806 513 17011 D AudioALSACaptureDataProviderBase: -attach()
12-31 20:14:32.806 513 17011 D AudioSPELayer: SetUPLinkDropTime, 0
12-31 20:14:32.807 513 17011 D AudioSPELayer: SetUPLinkIntrStartTime, sec=969, nsec=623045211
12-31 20:14:32.807 513 17012 D AudioALSACaptureDataProviderNormal: +readThread(), pid: 513, tid: 17012, kReadBufferSize=0x1e00, open_index=2
12-31 20:14:32.807 513 17011 D AudioALSACaptureDataClient: sample_rate: 48000 => 8000, num_channels: 2 => 2, audio_format: 0x4 => 0x1
12-31 20:14:32.807 513 17011 D MtkAudioSrc: MtkAudioSrc Constructor in SR 48000, CH 2; out SR 8000, CH 2; format 0
12-31 20:14:32.807 513 17011 D MtkAudioSrc: +open()
12-31 20:14:32.807 513 17011 D MtkAudioSrc: -open()
12-31 20:14:32.807 513 17011 D AudioALSACaptureDataClient: CheckNeedBesRecordSRC(), mStreamAttributeSource->sample_rate=48000, mStreamAttributeSource->num_channels=2, mStreamAttributeTarget->sample_rate=8000,mStreamAttributeTarget->num_channels=1, BesRecord_usingsamplerate=48000
12-31 20:14:32.807 513 17011 D MtkAudioSrc: MtkAudioSrc Constructor in SR 48000, CH 2; out SR 8000, CH 2; format 0
12-31 20:14:32.807 513 17011 D MtkAudioSrc: +open()
12-31 20:14:32.808 513 17011 D MtkAudioSrc: -open()
12-31 20:14:32.808 513 17011 D AudioALSACaptureDataClient: CheckNeedBesRecordSRC(), 0, 48000, mBesRecSRCSizeFactor=2
12-31 20:14:32.808 513 17011 D AudioALSACaptureDataClient: CheckChannelRemixOp(), ch 2->1 (mBesRecordStereoMode = 0, BesRecord enable = 1), ChannelRemixOp = 4
12-31 20:14:32.808 513 17011 D AudioSpeechEnhanceInfo: GetDebugStatus, mDebugflag=0
12-31 20:14:32.808 513 17011 D AudioALSAHardwareResourceManager: winsn:setInNormalRecord(), state=1
12-31 20:14:32.808 513 17011 D AudioALSAHardwareResourceManager: +startInputDevice(), new_device = 0x80000004, mInputDevice = 0x0, mStartInputDeviceCount = 0, mMicInverse=0
12-31 20:14:32.809 513 17011 D AudioALSAHardwareResourceManager: setMIC1Mode(), isphonemic 0, micmode 1
12-31 20:14:32.809 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceSettingByName() DeviceName = Mic1TypeACCMode descriptor->DeviceStatusCounte = 0
12-31 20:14:32.809 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_MIC1_Mode_Select cltvalue = ACCMODE
12-31 20:14:32.809 513 17011 D AudioALSAHardwareResourceManager: setMIC2Mode(), isphonemic 0, micmode 1
12-31 20:14:32.809 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceSettingByName() DeviceName = Mic2TypeACCMode descriptor->DeviceStatusCounte = 0
12-31 20:14:32.809 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_MIC2_Mode_Select cltvalue = ACCMODE
12-31 20:14:32.809 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnonSequenceByName() DeviceName = builtin_Mic_SingleMic descriptor->DeviceStatusCounte = 0
12-31 20:14:32.810 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_MicSource1_Setting cltvalue = ADC1
12-31 20:14:32.810 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnonSequence cltname.string () = Audio_MicSource1_Setting cltvalue.string () = ADC1
12-31 20:14:32.810 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_ADC_1_Switch cltvalue = On
12-31 20:14:32.810 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnonSequence cltname.string () = Audio_ADC_1_Switch cltvalue.string () = On
12-31 20:14:32.810 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_Preamp1_Switch cltvalue = IN_ADC1
12-31 20:14:32.811 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnonSequence cltname.string () = Audio_Preamp1_Switch cltvalue.string () = IN_ADC1
12-31 20:14:32.811 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_Preamp2_Switch cltvalue = IN_ADC1
12-31 20:14:32.811 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnonSequence cltname.string () = Audio_Preamp2_Switch cltvalue.string () = IN_ADC1
12-31 20:14:32.811 513 17011 D AudioALSAHardwareResourceManager: -startInputDevice(), mInputDevice = 0x80000004, mStartInputDeviceCount = 1
12-31 20:14:32.812 513 17011 D AudioVoiceUIDL: [SetUplinkStartTime] mULtime sec 0 nsec 0
12-31 20:14:32.812 513 17011 D AudioALSACaptureHandlerNormal: -open()
12-31 20:14:32.868 513 17011 D AudioVoiceUIDL: [SetUplinkStartTime] mULtime sec 969 nsec 634113903
12-31 20:14:41.577 14860 17009 D AudioRecord: stop
12-31 20:14:41.579 14860 17009 D AudioTrackShared: this(0x77f822c0e0), mCblk(0x77f76019c0), front(69632), mIsOut 0, interrupt() FUTEX_WAKE
12-31 20:14:41.587 513 17011 D AudioALSAStreamIn: +standby()
12-31 20:14:41.587 513 17011 D AudioALSAStreamIn: standby(), keep the mStandbyFrameCount = 70240, ret = 0
12-31 20:14:41.588 513 17011 D AudioALSAStreamIn: close()
12-31 20:14:41.588 513 17011 D AudioALSACaptureHandlerNormal: +close()
12-31 20:14:41.588 513 17011 D AudioALSAHardwareResourceManager: +stopInputDevice(), mInputDevice = 0x80000004, stop_device = 0x80000004, mStartInputDeviceCount = 1, mMicInverse=0
12-31 20:14:41.588 513 17011 D AudioALSAHardwareResourceManager: setMIC1Mode(), isphonemic 0, micmode 1
12-31 20:14:41.588 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceSettingByName() DeviceName = Mic1TypeACCMode descriptor->DeviceStatusCounte = 0
12-31 20:14:41.588 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_MIC1_Mode_Select cltvalue = ACCMODE
12-31 20:14:41.588 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnoffSequenceByName() DeviceName = builtin_Mic_SingleMic descriptor->DeviceStatusCounte = 1
12-31 20:14:41.588 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_Preamp1_Switch cltvalue = OPEN
12-31 20:14:41.588 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnoffSequence cltname.string () = Audio_Preamp1_Switch cltvalue.string () = OPEN
12-31 20:14:41.588 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_Preamp2_Switch cltvalue = OPEN
12-31 20:14:41.588 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnoffSequence cltname.string () = Audio_Preamp2_Switch cltvalue.string () = OPEN
12-31 20:14:41.589 513 17011 D AudioALSADeviceConfigManager: cltname = Audio_ADC_1_Switch cltvalue = Off
12-31 20:14:41.589 513 17011 D AudioALSADeviceConfigManager: ApplyDeviceTurnoffSequence cltname.string () = Audio_ADC_1_Switch cltvalue.string () = Off
12-31 20:14:41.589 513 17011 D AudioALSAHardwareResourceManager: -stopInputDevice(), mInputDevice = 0x0, mStartInputDeviceCount = 0
12-31 20:14:41.589 513 17011 D AudioALSACaptureDataClient: ~AudioALSACaptureDataClient()
12-31 20:14:41.589 513 17011 D AudioALSACaptureDataProviderBase: detach(),0xe6699000, Identity=1, mCaptureDataClientVector.size()=1,mCaptureDataProviderType=0, 0xe630c000
12-31 20:14:41.589 513 17011 D AudioALSACaptureDataProviderNormal: close()
12-31 20:14:41.611 513 17011 D AudioALSACaptureDataProviderNormal: -close()
12-31 20:14:41.612 513 17011 D AudioALSACaptureDataProviderBase: detach(), close finish
12-31 20:14:41.612 513 17011 D AudioALSACaptureDataProviderBase: -detach()
12-31 20:14:41.612 513 17011 D MtkAudioSrc: +close()
12-31 20:14:41.612 513 17012 D AudioALSACaptureDataProviderNormal: -readThread(), pid: 513, tid: 17012
12-31 20:14:41.612 513 17011 D MtkAudioSrc: -close()
12-31 20:14:41.612 513 17011 D MtkAudioSrc: +~MtkAudioSrc()
12-31 20:14:41.612 513 17011 D MtkAudioSrc: -~MtkAudioSrc()
12-31 20:14:41.612 513 17011 D AudioALSACaptureDataClient: +StopBesRecord()
12-31 20:14:41.612 513 17011 D AudioSPELayer: Stop
12-31 20:14:41.612 513 17011 D AudioSPELayer: Clear
12-31 20:14:41.612 513 17011 D AudioSPELayer: free mSphCtrlBuffer 0xe6349d80
12-31 20:14:41.612 513 17011 D AudioSPELayer: ~free mSphCtrlBuffer
12-31 20:14:41.612 513 17011 D AudioSPELayer: ~Clear
12-31 20:14:41.612 513 17011 D AudioALSACaptureDataClient: -StopBesRecord()
12-31 20:14:41.612 513 17011 D MtkAudioSrc: +close()
12-31 20:14:41.612 513 17011 D MtkAudioSrc: -close()
12-31 20:14:41.612 513 17011 D MtkAudioSrc: +~MtkAudioSrc()
12-31 20:14:41.613 513 17011 D MtkAudioSrc: -~MtkAudioSrc()
12-31 20:14:41.613 513 17011 D AudioSPELayer: ~SPELayer+
12-31 20:14:41.613 513 17011 D AudioSPELayer: Clear
12-31 20:14:41.613 513 17011 D AudioSPELayer: ~Clear
12-31 20:14:41.613 513 17011 D AudioSPELayer: FlushBufferQ+
12-31 20:14:41.613 513 17011 D AudioSPELayer: FlushBufferQ mULOutBufferQ size=1,mULInBufferQ.size=1,mDLOutBufferQ.size()=0,mDLInBufferQ.size()=0,mDLDelayBufferQ.size()=0
12-31 20:14:41.613 513 17011 D AudioSPELayer: FlushBufferQ-
12-31 20:14:41.614 513 17011 D AudioSPELayer: ~SPELayer-
12-31 20:14:41.614 513 17011 D AudioPreProcess: ~AudioPreProcess()+
12-31 20:14:41.614 513 17011 D AudioPreProcess: ~AudioPreProcess()-
12-31 20:14:41.614 513 17011 D AudioALSACaptureDataClient: -~AudioALSACaptureDataClient()
12-31 20:14:41.614 513 17011 D AudioALSAHardwareResourceManager: winsn:setInNormalRecord(), state=0
12-31 20:14:41.614 513 17011 D AudioVoiceUIDL: [SetUplinkStartTime] mULtime sec 0 nsec 0
12-31 20:14:41.614 513 17011 D AudioALSACaptureHandlerNormal: -close()
12-31 20:14:41.614 513 17011 D AudioALSAStreamManager: +destroyCaptureHandler(), mode = 0, pCaptureHandler = 0xe630f040
12-31 20:14:41.614 513 17011 D AudioALSACaptureHandlerNormal: +~AudioALSACaptureHandlerNormal()
12-31 20:14:41.614 513 17011 D AudioALSACaptureHandlerNormal: ~AudioALSACaptureHandlerNormal()
12-31 20:14:41.614 513 17011 D AudioALSACaptureHandlerBase: ~AudioALSACaptureHandlerBase()
12-31 20:14:41.615 513 17011 D AudioALSAStreamManager: -destroyCaptureHandler(), mCaptureHandlerVector.size() = 0
12-31 20:14:41.615 513 17011 D AudioALSAStreamIn: -standby()
12-31 20:14:41.615 513 1349 V APM_AudioPolicyManager: stopInput() input 38
12-31 20:14:41.615 513 1349 V AudioPolicyService: AudioCommandThread() adding recording configuration update event 0, source 1
12-31 20:14:41.615 513 1349 V AudioPolicyService: AudioCommandThread() adding release patch delay 0
12-31 20:14:41.615 513 794 V AudioPolicyService: AudioCommandThread() processing recording configuration update
12-31 20:14:41.615 513 793 V AudioPolicyService: AudioCommandThread() processing release audio patch
12-31 20:14:41.616 513 17011 D AudioALSAHardware: -releaseAudioPatch handle [0x4]
12-31 20:14:41.616 513 17011 D AudioALSAHardware: +routing releaseAudioPatch 80000004->Mixer
12-31 20:14:41.616 513 17011 D AudioALSAStreamManager: +setParameters(), IOport = 38, keyValuePairs = routing=0
12-31 20:14:41.616 513 17011 D AudioALSAStreamManager: Send to mStreamInVector [1]
12-31 20:14:41.616 513 17011 D AudioALSAStreamIn: +setParameters(): routing=0
12-31 20:14:41.616 513 17011 D AudioALSAStreamManager: +routingInputDevice(), input_device: 0x80000004 => 0x0
12-31 20:14:41.616 513 17011 W AudioALSAStreamManager: -routingInputDevice(), input_device == AUDIO_DEVICE_NONE(0x0), return
12-31 20:14:41.616 513 17011 D AudioALSAStreamIn: -setParameters(): routing=0
12-31 20:14:41.617 513 17011 D AudioALSAStreamManager: -setParameters()
12-31 20:14:41.617 513 17011 D AudioALSAHardware: handlecheck releaseAudioPatch remove handle [4] OK
12-31 20:14:41.617 513 17011 D AudioALSAHardware: -releaseAudioPatch handle [0x4] status [0]
12-31 20:14:41.620 513 1349 V APM_AudioPolicyManager: resetInputDevice() releaseAudioPatch returned 0
12-31 20:14:41.621 513 1349 V APM_AudioPolicyManager: removeAudioPatch() handle 4 af handle 36
12-31 20:14:41.623 14860 17009 D AudioRecord: -stop
