Android音频系统探究——从SoundPool到AudioHardware
分类:android 2014-08-08 14:43 316人阅读 评论(0) 收藏 举报 对音频系统的探索起源于工作中遇到的一个bug。平时都是力求快速解决问题,不问原因。这次时间比较宽裕,正好借着解决问题的机会,把Android的音频系统了解一下。既然由bug引发,那就从bug开始说。
一. bug现象
Android的照相机在拍照的时候会播放一个按键音。最近的一个MID项目(基于RK3188,Android 4.2)中,测试部门反馈,拍照时按键音播放异常情况如下:
(1)进入应用程序以后,第一次拍照,没有按键音
(2)连续拍照,有按键音
(3)停止连拍,等待几秒钟后,再次拍照,又没有按键音
二. 问题简化
看CameraApp代码可以知道,播放按键音使用了SoundPool类。做一个使用SoundPool播放声音的应用程序,界面上只有一个Button,点击后播放声音。这样就能确定这单纯是声音播放问题还是复合性问题。代码很简单:
- protectedvoidonCreate(BundlesavedInstanceState){
- super.onCreate(savedInstanceState);
- setContentView(R.layout.activity_test_sound_pool);
- mSoundPool=newSoundPool(10,AudioManager.STREAM_SYSTEM,5);
- mSoundId=mSoundPool.load(this,R.raw.camera_click,1);//这里R.raw.camera_click是ogg格式的音频资源
- vBtnShut=(Button)findViewById(R.id.btn_click);
- vBtnShut.setOnClickListener(newOnClickListener(){
- @Override
- publicvoidonClick(Viewv){
- mSoundPool.play(mSoundId,1,1,0,0,1);
- }
- });
- }
- protectedvoidonCreate(BundlesavedInstanceState){
- super.onCreate(savedInstanceState);
- setContentView(R.layout.activity_test_sound_pool);
- mSoundPool=newSoundPool(10,AudioManager.STREAM_SYSTEM,5);
- mSoundId=mSoundPool.load(this,R.raw.camera_click,1);//这里R.raw.camera_click是ogg格式的音频资源
- vBtnShut=(Button)findViewById(R.id.btn_click);
- vBtnShut.setOnClickListener(newOnClickListener(){
- @Override
- publicvoidonClick(Viewv){
- mSoundPool.play(mSoundId,1,1,0,0,1);
- }
- });
- }
idle-->play failed-->idle-->play failed-->play success-->play success-->idle-->play failed-->...
可以总结为,每间隔几秒钟后,第一次播放音频无声音输出。
三. 初步分析
理清了现象,简化了环境,我们可以开始分析问题了:
显而易见的是,BUG非常规律,只有相隔几秒钟后的第一次播放才出现问题,与软件逻辑密切相关,可以排除硬件问题。本质上来讲,无论使用什么软件系统,声音播放的流程一般都是——用户指定要播放的声音数据,可能是文件,可能是Buffer;Audio系统对声音数据解码,可能采用软解码,也可能采用硬解码;将解码出来的数字音频信号传给功放设备,经过D/A转换后送到扬声器,声音就播放出来了。可以说,这个流程中的第一部分,是应用程序的行为;第二部分,是Android系统的职责;第三部分,是kernel中驱动的工作。应用程序的问题可以排除,现在要解决的疑问是,是解码程序出了问题,还是驱动程序出了问题?出现了什么情况,导致了idle后播放不出来?
四. 代码研究
1. Android Audio框架
首先网络上找找资料,要搞清楚Android音频的框架层次结构,才容易定位问题。用图说明——
有了大致的概念,开始以SoundPool为入口,摸清播放流程。其中在每个层次中要了解两点:数据如何传递,播放的动作如何执行。 也就是沿着SoundPool.load()和Sound.play()顺藤摸瓜。
2. SoundPool和AudioFlinger
SoundPool.java基本是个空壳,直接使用了Native接口,代码没什么可看的。不过可以先看下这个类的介绍,就在SoundPool.java的开头,整一页的英文注释。幸运的是,很快就找到了我们需要看的资料:
[cpp] view plain copy- /**
- *TheSoundPoolclassmanagesandplaysaudioresourcesforapplications.
- *
- *<p>ASoundPoolisacollectionofsamplesthatcanbeloadedintomemory
- *fromaresourceinsidetheAPKorfromafileinthefilesystem.The
- *SoundPoollibraryusestheMediaPlayerservicetodecodetheaudio
- *intoaraw16-bitPCMmonoorstereostream.Thisallowsapplications
- *toshipwithcompressedstreamswithouthavingtosuffertheCPUload
- *andlatencyofdecompressingduringplayback.</p>
- ......
- ......
- */
- /**
- *TheSoundPoolclassmanagesandplaysaudioresourcesforapplications.
- *
- *<p>ASoundPoolisacollectionofsamplesthatcanbeloadedintomemory
- *fromaresourceinsidetheAPKorfromafileinthefilesystem.The
- *SoundPoollibraryusestheMediaPlayerservicetodecodetheaudio
- *intoaraw16-bitPCMmonoorstereostream.Thisallowsapplications
- *toshipwithcompressedstreamswithouthavingtosuffertheCPUload
- *andlatencyofdecompressingduringplayback.</p>
- ......
- ......
- */
挑重要的说,SoundPool是Sample的集合,能把APK里的资源或者文件系统中的文件加载到内存中,使用MediaPlayer服务把音频解码成原始的16位PCM单声道或立体声数据流。好嘛,原来解码在这里就做了。还是看看代码实现吧,免得心里不踏实。
不去理会Jni那些手续,直接看SoundPool.cpp。上面那个测试APK的代码,调用了SoundPool的load,play两个接口,就把声音播放出来了。load一次后,可多次播放,这两个接口之所以要分开,应该就是load做了解码。先看load的实现,为满足不同音频资源的需要,load被重载了,看其中一个就行了。
[cpp] view plain copy- intSoundPool::load(intfd,int64_toffset,int64_tlength,intpriority)
- {
- ALOGV("load:fd=%d,offset=%lld,length=%lld,priority=%d",
- fd,offset,length,priority);
- Mutex::Autolocklock(&mLock);
- sp<Sample>sample=newSample(++mNextSampleID,fd,offset,length);
- mSamples.add(sample->sampleID(),sample);//将sample对象加入管理
- doLoad(sample);//load所在
- returnsample->sampleID();
- }
- intSoundPool::load(intfd,int64_toffset,int64_tlength,intpriority)
- {
- ALOGV("load:fd=%d,offset=%lld,length=%lld,priority=%d",
- fd,offset,length,priority);
- Mutex::Autolocklock(&mLock);
- sp<Sample>sample=newSample(++mNextSampleID,fd,offset,length);
- mSamples.add(sample->sampleID(),sample);//将sample对象加入管理
- doLoad(sample);//load所在
- returnsample->sampleID();
- }
数据处理角度来说,真正的load在doLoad中:
[cpp] view plain copy- voidSoundPool::doLoad(sp<Sample>&sample)
- {
- ALOGV("doLoad:loadingsamplesampleID=%d",sample->sampleID());
- sample->startLoad();//只是改变了状态
- mDecodeThread->loadSample(sample->sampleID());//真正加载的地方
- }
- voidSoundPool::doLoad(sp<Sample>&sample)
- {
- ALOGV("doLoad:loadingsamplesampleID=%d",sample->sampleID());
- sample->startLoad();//只是改变了状态
- mDecodeThread->loadSample(sample->sampleID());//真正加载的地方
- }
看到了mDecodeThread,眼前一亮,很可能这里就是将ogg解码成PCM的地方了。所以进入loadSample看一看: [cpp] view plain copy
- voidSoundPoolThread::loadSample(intsampleID){
- write(SoundPoolMsg(SoundPoolMsg::LOAD_SAMPLE,sampleID));
- }
- voidSoundPoolThread::loadSample(intsampleID){
- write(SoundPoolMsg(SoundPoolMsg::LOAD_SAMPLE,sampleID));
- }
只是消息传递而已,找到LOAD_SAMPLE消息处理的地方:
[cpp] view plain copy- intSoundPoolThread::run(){
- ALOGV("run");
- for(;;){
- SoundPoolMsgmsg=read();
- ALOGV("Gotmessagem=%d,mData=%d",msg.mMessageType,msg.mData);
- switch(msg.mMessageType){
- caseSoundPoolMsg::KILL:
- ALOGV("goodbye");
- returnNO_ERROR;
- caseSoundPoolMsg::LOAD_SAMPLE://在这里处理LOAD_SAMPLE
- doLoadSample(msg.mData);
- break;
- default:
- ALOGW("run:Unrecognizedmessage%d\n",
- msg.mMessageType);
- break;
- }
- }
- }
- intSoundPoolThread::run(){
- ALOGV("run");
- for(;;){
- SoundPoolMsgmsg=read();
- ALOGV("Gotmessagem=%d,mData=%d",msg.mMessageType,msg.mData);
- switch(msg.mMessageType){
- caseSoundPoolMsg::KILL:
- ALOGV("goodbye");
- returnNO_ERROR;
- caseSoundPoolMsg::LOAD_SAMPLE://在这里处理LOAD_SAMPLE
- doLoadSample(msg.mData);
- break;
- default:
- ALOGW("run:Unrecognizedmessage%d\n",
- msg.mMessageType);
- break;
- }
- }
- }
- voidSoundPoolThread::doLoadSample(intsampleID){
- sp<Sample>sample=mSoundPool->findSample(sampleID);
- status_tstatus=-1;
- if(sample!=0){
- status=sample->doLoad();
- }
- mSoundPool->notify(SoundPoolEvent(SoundPoolEvent::SAMPLE_LOADED,sampleID,status));
- }
- voidSoundPoolThread::doLoadSample(intsampleID){
- sp<Sample>sample=mSoundPool->findSample(sampleID);
- status_tstatus=-1;
- if(sample!=0){
- status=sample->doLoad();
- }
- mSoundPool->notify(SoundPoolEvent(SoundPoolEvent::SAMPLE_LOADED,sampleID,status));
- }
看来最后是在sample->doLoad()中做的处理。进去看看,颇有惊喜:
[cpp] view plain copy- status_tSample::doLoad()
- {
- uint32_tsampleRate;
- intnumChannels;
- audio_format_tformat;
- sp<IMemory>p;
- ALOGV("Startdecode");
- if(mUrl){
- p=MediaPlayer::decode(mUrl,&sampleRate,&numChannels,&format);
- }else{
- p=MediaPlayer::decode(mFd,mOffset,mLength,&sampleRate,&numChannels,&format);
- ALOGV("close(%d)",mFd);
- ::close(mFd);
- mFd=-1;
- }
- if(p==0){
- ALOGE("Unabletoloadsample:%s",mUrl);
- return-1;
- }
- ALOGV("pointer=%p,size=%u,sampleRate=%u,numChannels=%d",
- p->pointer(),p->size(),sampleRate,numChannels);
- if(sampleRate>kMaxSampleRate){
- ALOGE("Samplerate(%u)outofrange",sampleRate);
- return-1;
- }
- if((numChannels<1)||(numChannels>2)){
- ALOGE("Samplechannelcount(%d)outofrange",numChannels);
- return-1;
- }
- //_dumpBuffer(p->pointer(),p->size());
- uint8_t*q=static_cast<uint8_t*>(p->pointer())+p->size()-10;
- //_dumpBuffer(q,10,10,false);
- mData=p;
- mSize=p->size();
- mSampleRate=sampleRate;
- mNumChannels=numChannels;
- mFormat=format;
- mState=READY;
- return0;
- }
- status_tSample::doLoad()
- {
- uint32_tsampleRate;
- intnumChannels;
- audio_format_tformat;
- sp<IMemory>p;
- ALOGV("Startdecode");
- if(mUrl){
- p=MediaPlayer::decode(mUrl,&sampleRate,&numChannels,&format);
- }else{
- p=MediaPlayer::decode(mFd,mOffset,mLength,&sampleRate,&numChannels,&format);
- ALOGV("close(%d)",mFd);
- ::close(mFd);
- mFd=-1;
- }
- if(p==0){
- ALOGE("Unabletoloadsample:%s",mUrl);
- return-1;
- }
- ALOGV("pointer=%p,size=%u,sampleRate=%u,numChannels=%d",
- p->pointer(),p->size(),sampleRate,numChannels);
- if(sampleRate>kMaxSampleRate){
- ALOGE("Samplerate(%u)outofrange",sampleRate);
- return-1;
- }
- if((numChannels<1)||(numChannels>2)){
- ALOGE("Samplechannelcount(%d)outofrange",numChannels);
- return-1;
- }
- //_dumpBuffer(p->pointer(),p->size());
- uint8_t*q=static_cast<uint8_t*>(p->pointer())+p->size()-10;
- //_dumpBuffer(q,10,10,false);
- mData=p;
- mSize=p->size();
- mSampleRate=sampleRate;
- mNumChannels=numChannels;
- mFormat=format;
- mState=READY;
- return0;
- }
弄清楚SoundPool的Play做了什么,也就能找到HAL的代码了。下面看只看play中的关键代码:
- intSoundPool::play(intsampleID,floatleftVolume,floatrightVolume,
- intpriority,intloop,floatrate)
- {
- //...
- channel=allocateChannel_l(priority);
- //...
- channel->play(sample,channelID,leftVolume,rightVolume,priority,loop,rate);
- //...
- }
- intSoundPool::play(intsampleID,floatleftVolume,floatrightVolume,
- intpriority,intloop,floatrate)
- {
- //...
- channel=allocateChannel_l(priority);
- //...
- channel->play(sample,channelID,leftVolume,rightVolume,priority,loop,rate);
- //...
- }
- voidSoundChannel::play(constsp<Sample>&sample,intnextChannelID,floatleftVolume,
- floatrightVolume,intpriority,intloop,floatrate)
- {
- AudioTrack*newTrack;
- //....
- newTrack=newAudioTrack(streamType,sampleRate,sample->format(),
- channels,frameCount,AUDIO_OUTPUT_FLAG_FAST,callback,userData,bufferFrames);
- //...
- mState=PLAYING;
- mAudioTrack->start();
- //...
- }
- voidSoundChannel::play(constsp<Sample>&sample,intnextChannelID,floatleftVolume,
- floatrightVolume,intpriority,intloop,floatrate)
- {
- AudioTrack*newTrack;
- //....
- newTrack=newAudioTrack(streamType,sampleRate,sample->format(),
- channels,frameCount,AUDIO_OUTPUT_FLAG_FAST,callback,userData,bufferFrames);
- //...
- mState=PLAYING;
- mAudioTrack->start();
- //...
- }
SoundChannel::play创建了一个AudioTrack对象,在AudioTrack的构造函数中,调用了set,set又调用了createTrack_l。createTrack_I中,通过IAudioFlinger创建了一个IAudioTrack。关于AudioTrack和AudioFlinger是为何物,两者如何交换音频数据,就说来话长了。而且有很多大大分析得很详细,就不赘述了。有几篇写得很好——
- AudioTrack分析:http://www.cnblogs.com/innost/archive/2011/01/09/1931457.html
- AudioFlinger分析:http://www.cnblogs.com/innost/archive/2011/01/15/1936425.html
- AudioTrack如何与AudioFlinger交换数据:http://blog.chinaunix.net/uid-26533928-id-3052398.html
阅读这些资料我们可以知道,Android Framework的音频子系统中,每一个音频流对应着一个AudioTrack类的一个实例,每个AudioTrack会在创建时注册到AudioFlinger中,由AudioFlinger把所有的AudioTrack进行混合(Mixer),然后输送到AudioHardware中进行播放。换言之,AudioFlinger是Audio系统的核心服务之一,起到了承上启下的衔接作用。
我们现在已经让SoundPool牵线,抓到AudioFlinger这条大鱼。下面着重来看AudioFlinger如何向下调用AudioHardware的。
3. AudioFlinger与AudioHardware
这里需要一点基础知识,先要了解Android的硬件抽象接口机制,才能理解AudioFlinger如何调用到AudioHardware,相关资料:
http://blog.csdn.net/myarrow/article/details/7175204
因为对Audio系统一无所知,所以很惭愧用了反相的代码搜索,在hardware/xxx/audio目录下查找HAL_MODULE_INFO_SYM,然后反过来到framework找HAL_MODULE_INFO_SYM的id "AUDIO_HARDWARE_MODULE_ID",过程非常笨拙,不足为道。他山之石可以攻玉,看到一篇好文,借助其中的一段分析来完成对AudioFlinger和AudioHardware关联的分析。原文地址:http://blog.csdn.net/xuesen_lin/article/details/8805108
当AudioPolicyService构造时创建了一个AudioPolicyDevice(mpAudioPolicyDev)并由此打开一个AudioPolicy(mpAudioPolicy)——这个Policy默认情况下的实现是legacy_audio_policy::policy(数据类型audio_policy)。同时legacy_audio_policy还包含了一个AudioPolicyInterface成员变量,它会被初始化为一个AudioPolicyManagerDefault。AudioPolicyManagerDefault的父类,即AudioPolicyManagerBase,它的构造函数中调用了mpClientInterface->loadHwModule()。
[cpp] view plain copy- AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface*clientInterface)…
- {
- //......
- for(size_ti=0;i<mHwModules.size();i++){
- mHwModules[i]->mHandle=mpClientInterface->loadHwModule(mHwModules[i]->mName);
- if(mHwModules[i]->mHandle==0){
- continue;
- }
- //......
- }
- AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface*clientInterface)…
- {
- //......
- for(size_ti=0;i<mHwModules.size();i++){
- mHwModules[i]->mHandle=mpClientInterface->loadHwModule(mHwModules[i]->mName);
- if(mHwModules[i]->mHandle==0){
- continue;
- }
- //......
- }
很明显的mpClientInterface这个变量在AudioPolicyManagerBase构造函数中做了初始化,再回溯追踪,可以发现它的根源在AudioPolicyService的构造函数中,对应的代码语句如下:
- rc=mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev,&aps_ops,this,&mpAudioPolicy);
- rc=mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev,&aps_ops,this,&mpAudioPolicy);
在这个场景下,函数create_audio_policy对应的是create_legacy_ap,并将传入的aps_ops组装到一个AudioPolicyCompatClient对象中,也就是mpClientInterface所指向的那个对象。
换句话说,mpClientInterface->loadHwModule实际上调用的就是aps_ops->loadHwModule,即:
- staticaudio_module_handle_taps_load_hw_module(void*service,constchar*name)
- {
- sp<IAudioFlinger>af=AudioSystem::get_audio_flinger();
- …
- returnaf->loadHwModule(name);
- }
- staticaudio_module_handle_taps_load_hw_module(void*service,constchar*name)
- {
- sp<IAudioFlinger>af=AudioSystem::get_audio_flinger();
- …
- returnaf->loadHwModule(name);
- }
AudioFlinger终于出现了,同样的情况也适用于mpClientInterface->openOutput,代码如下:
[cpp] view plain copy
- staticaudio_io_handle_taps_open_output(…)
- {
- sp<IAudioFlinger>af=AudioSystem::get_audio_flinger();
- …
- returnaf->openOutput((audio_module_handle_t)0,pDevices,pSamplingRate,pFormat,pChannelMask,
- pLatencyMs,flags);
- }
- staticaudio_io_handle_taps_open_output(…)
- {
- sp<IAudioFlinger>af=AudioSystem::get_audio_flinger();
- …
- returnaf->openOutput((audio_module_handle_t)0,pDevices,pSamplingRate,pFormat,pChannelMask,
- pLatencyMs,flags);
- }
现在前方就是AudioHardware了,终于打开了从APK到HAL的通路。
4. AudioHardware
AudioHardware有两个内部类,AudioStreamOutALSA和AudioStreamInALSA,我们要解决的是声音播放的问题,看AudioStreamOutALSA即可。 AudioStreamOutALSA代码很清晰,很快找到了我们需要的代码,写PCM数据用的函数:
[cpp] view plain copy- AudioHardware::AudioStreamOutALSA::AudioStreamOutALSA():
- mHardware(0),mPcm(0),mMixer(0),mRouteCtl(0),
- mStandby(true),mDevices(0),mChannels(AUDIO_HW_OUT_CHANNELS),
- mSampleRate(AUDIO_HW_OUT_SAMPLERATE),mBufferSize(AUDIO_HW_OUT_PERIOD_BYTES),
- mDriverOp(DRV_NONE),mStandbyCnt(0)
- {
- #ifdefDEBUG_ALSA_OUT
- if(alsa_out_fp==NULL)
- alsa_out_fp=fopen("/data/data/out.pcm","a+");
- if(alsa_out_fp)
- ALOGI("------------>openfilesuccess");
- #endif
- }
- ssize_tAudioHardware::AudioStreamOutALSA::write(constvoid*buffer,size_tbytes)
- {
- //...
- #ifdefDEBUG_ALSA_OUT
- if(alsa_out_fp)
- fwrite(buffer,1,bytes,alsa_out_fp);
- #endif
- //...
- if(mStandby){
- open_l();//重新open音频设备
- mStandby=false;
- }
- //...
- ret=pcm_write(mPcm,(void*)p,bytes);
- //...
- }
- AudioHardware::AudioStreamOutALSA::AudioStreamOutALSA():
- mHardware(0),mPcm(0),mMixer(0),mRouteCtl(0),
- mStandby(true),mDevices(0),mChannels(AUDIO_HW_OUT_CHANNELS),
- mSampleRate(AUDIO_HW_OUT_SAMPLERATE),mBufferSize(AUDIO_HW_OUT_PERIOD_BYTES),
- mDriverOp(DRV_NONE),mStandbyCnt(0)
- {
- #ifdefDEBUG_ALSA_OUT
- if(alsa_out_fp==NULL)
- alsa_out_fp=fopen("/data/data/out.pcm","a+");
- if(alsa_out_fp)
- ALOGI("------------>openfilesuccess");
- #endif
- }
- ssize_tAudioHardware::AudioStreamOutALSA::write(constvoid*buffer,size_tbytes)
- {
- //...
- #ifdefDEBUG_ALSA_OUT
- if(alsa_out_fp)
- fwrite(buffer,1,bytes,alsa_out_fp);
- #endif
- //...
- if(mStandby){
- open_l();//重新open音频设备
- mStandby=false;
- }
- //...
- ret=pcm_write(mPcm,(void*)p,bytes);
- //...
- }
这里提供了一个很容易验证PCM数据是否正确的方法,打开DEBUG_ALSA_OUT开关后,可以将PCM流保存到“/data/data/out.pcm”文件中。到了验证数据是否正确的时候了,打开这个编译开关,得到out.pcm,将它pull到PC上,用coolEdit打开播放,发现正常播放正常。好了,我们现在可以知道,问题并不出在解码程序上了。那又是什么原因导致的呢,我们从write函数开始,研究播放的流程。
首先,AudioStreamOutALSA的构造函数中将mStandby初始化为true。这个变量显然是作为记录音频设备待机状态用的。当mStandby==true时,每次调用write,都会调用open_l()重新开启一次音频设备,然后再做pcm_write。
再看看open_l():
[cpp] view plain copy- status_tAudioHardware::AudioStreamOutALSA::open_l()
- {
- //...
- mPcm=mHardware->openPcmOut_l();
- if(mPcm==NULL){
- returnNO_INIT;
- }
- //...
- }
- structpcm*AudioHardware::openPcmOut_l()
- {
- //...
- mPcm=pcm_open(flags);
- //...
- if(!pcm_ready(mPcm)){
- pcm_close(mPcm);
- //...
- }
- }
- returnmPcm;
- }
- status_tAudioHardware::AudioStreamOutALSA::open_l()
- {
- //...
- mPcm=mHardware->openPcmOut_l();
- if(mPcm==NULL){
- returnNO_INIT;
- }
- //...
- }
- structpcm*AudioHardware::openPcmOut_l()
- {
- //...
- mPcm=pcm_open(flags);
- //...
- if(!pcm_ready(mPcm)){
- pcm_close(mPcm);
- //...
- }
- }
- returnmPcm;
- }
- structpcm*pcm_open(unsignedflags)
- {
- //......
- if(flags&PCM_IN){
- dname="/dev/snd/pcmC0D0c";
- channalFlags=-1;
- startCheckCount=0;
- }else{
- #ifdefSUPPORT_USB
- dname="/dev/snd/pcmC1D0p";
- #else
- dname="/dev/snd/pcmC0D0p";
- #endif
- }
- pcm->fd=open(dname,O_RDWR);
- if(pcm->fd<0){
- oops(pcm,errno,"cannotopendevice'%s'",dname);
- returnpcm;
- }
- if(ioctl(pcm->fd,SNDRV_PCM_IOCTL_INFO,&info)){
- oops(pcm,errno,"cannotgetinfo-%s",dname);
- gotofail;
- }
- if(ioctl(pcm->fd,SNDRV_PCM_IOCTL_HW_PARAMS,¶ms)){
- oops(pcm,errno,"cannotsethwparams");
- gotofail;
- }
- if(ioctl(pcm->fd,SNDRV_PCM_IOCTL_SW_PARAMS,&sparams)){
- oops(pcm,errno,"cannotsetswparams");
- gotofail;
- }
- fail:
- close(pcm->fd);
- pcm->fd=-1;
- returnpcm;
- }
- structpcm*pcm_open(unsignedflags)
- {
- //......
- if(flags&PCM_IN){
- dname="/dev/snd/pcmC0D0c";
- channalFlags=-1;
- startCheckCount=0;
- }else{
- #ifdefSUPPORT_USB
- dname="/dev/snd/pcmC1D0p";
- #else
- dname="/dev/snd/pcmC0D0p";
- #endif
- }
- pcm->fd=open(dname,O_RDWR);
- if(pcm->fd<0){
- oops(pcm,errno,"cannotopendevice'%s'",dname);
- returnpcm;
- }
- if(ioctl(pcm->fd,SNDRV_PCM_IOCTL_INFO,&info)){
- oops(pcm,errno,"cannotgetinfo-%s",dname);
- gotofail;
- }
- if(ioctl(pcm->fd,SNDRV_PCM_IOCTL_HW_PARAMS,¶ms)){
- oops(pcm,errno,"cannotsethwparams");
- gotofail;
- }
- if(ioctl(pcm->fd,SNDRV_PCM_IOCTL_SW_PARAMS,&sparams)){
- oops(pcm,errno,"cannotsetswparams");
- gotofail;
- }
- fail:
- close(pcm->fd);
- pcm->fd=-1;
- returnpcm;
- }
果然,这里就是操作设备节点的地方了。我们先在AudioStreamOutALSA的write中加打印信息,看看第一次播放和后续播放究竟有何不同。测试结果发现,每次播放不出声音的情况,都发生mStandby==true之后,这个时候做了一次打开音频设备的动作,但此时PCM数据是正确的。我们先来看看什么时候会导致mStandby==true。 [cpp] view plain copy
- <PREclass=cppname="code"><PREclass=cppname="code">status_tAudioHardware::AudioStreamOutALSA::standby()
- {
- doStandby_l();
- }
- voidAudioHardware::AudioStreamOutALSA::doStandby_l()
- {
- if(!mStandby)
- mStandby=true;
- close_l();
- }
- voidAudioHardware::AudioStreamOutALSA::close_l()
- {
- if(mPcm){
- mHardware->closePcmOut_l();
- mPcm=NULL;
- }
- }</PRE><BR><BR></PRE>
- <divclass="dp-highlighterbg_cpp"><divclass="bar"><divclass="tools"><strong>[cpp]</strong><atarget=_blankclass="ViewSource"title="viewplain"href="http://blog.csdn.net/special_lin/article/details/12849637#">viewplain</a><atarget=_blankclass="CopyToClipboard"title="copy"href="http://blog.csdn.net/special_lin/article/details/12849637#">copy</a><atarget=_blankclass="PrintSource"title="print"href="http://blog.csdn.net/special_lin/article/details/12849637#">print</a><atarget=_blankclass="About"title="?"href="http://blog.csdn.net/special_lin/article/details/12849637#">?</a></div></div><olclass="dp-cpp"><liclass="alt"><span><span><PRE</span><spanclass="keyword">class</span><span>=cppname=</span><spanclass="string">"code"</span><span>>status_tAudioHardware::AudioStreamOutALSA::standby()</span></span></li><li><span>{</span></li><liclass="alt"><span>doStandby_l();</span></li><li><span>}</span></li><liclass="alt"><span></span></li><li><span></span><spanclass="keyword">void</span><span>AudioHardware::AudioStreamOutALSA::doStandby_l()</span></li><liclass="alt"><span>{</span></li><li><span></span></li><liclass="alt"><span></span><spanclass="keyword">if</span><span>(!mStandby)</span></li><li><span>mStandby=</span><spanclass="keyword">true</span><span>;</span></li><liclass="alt"><span>close_l();</span></li><li><span>}</span></li><liclass="alt"><span></span></li><li><span></span><spanclass="keyword">void</span><span>AudioHardware::AudioStreamOutALSA::close_l()</span></li><liclass="alt"><span>{</span></li><li><span></span><spanclass="keyword">if</span><span>(mPcm){</span></li><liclass="alt"><span>mHardware->closePcmOut_l();</span></li><li><span>mPcm=NULL;</span></li><liclass="alt"><span>}</span></li><li><span>}</PRE><BR><BR></span></li></ol></div><prestyle="DISPLAY:none"class="cpp"name="code"><divclass="dp-highlighterbg_cpp"><divclass="bar"><divclass="tools"><strong>[cpp]</strong><atarget=_blankclass="ViewSource"title="viewplain"href="http://blog.csdn.net/special_lin/article/details/12849637#">viewplain</a><atarget=_blankclass="CopyToClipboard"title="copy"href="http://blog.csdn.net/special_lin/article/details/12849637#">copy</a><atarget=_blankclass="PrintSource"title="print"href="http://blog.csdn.net/special_lin/article/details/12849637#">print</a><atarget=_blankclass="About"title="?"href="http://blog.csdn.net/special_lin/article/details/12849637#">?</a></div></div><olclass="dp-cpp"><liclass="alt"><span><span>status_tAudioHardware::AudioStreamOutALSA::standby()</span></span></li><li><span>{</span></li><liclass="alt"><span>doStandby_l();</span></li><li><span>}</span></li><liclass="alt"><span></span></li><li><span></span><spanclass="keyword">void</span><span>AudioHardware::AudioStreamOutALSA::doStandby_l()</span></li><liclass="alt"><span>{</span></li><li><span></span></li><liclass="alt"><span></span><spanclass="keyword">if</span><span>(!mStandby)</span></li><li><span>mStandby=</span><spanclass="keyword">true</span><span>;</span></li><liclass="alt"><span>close_l();</span></li><li><span>}</span></li><liclass="alt"><span></span></li><li><span></span><spanclass="keyword">void</span><span>AudioHardware::AudioStreamOutALSA::close_l()</span></li><liclass="alt"><span>{</span></li><li><span></span><spanclass="keyword">if</span><span>(mPcm){</span></li><liclass="alt"><span>mHardware->closePcmOut_l();</span></li><li><span>mPcm=NULL;</span></li><liclass="alt"><span>}</span></li><li><span>}</span></li></ol></div><prestyle="DISPLAY:none"class="cpp"name="code">status_tAudioHardware::AudioStreamOutALSA::standby()
- {
- doStandby_l();
- }
- voidAudioHardware::AudioStreamOutALSA::doStandby_l()
- {
- if(!mStandby)
- mStandby=true;
- close_l();
- }
- voidAudioHardware::AudioStreamOutALSA::close_l()
- {
- if(mPcm){
- mHardware->closePcmOut_l();
- mPcm=NULL;
- }
- }
好了,现在我们可以确定,mStandby是在调用standby的时候被设置生true了。如果不总是重新打开音频设备,会不会变正常?做了一个实验,把standby函数体的代码都注释掉。这样修改后,果然开机只有一次声音播放不出来,那就是第一次。每隔一段时间,声音就播不出来的问题不见了。
其实到现在,问题已经定位出来了。这个问题属于kernel问题,不再属于Framework了。但是还是想弄清楚,standby为什么隔一段时间被调用一次,是被谁调用的。经过一系列反查,找到了standby的真正调用处,AudioFlinger的播放线程中。具体怎么查的,还是要参考HAL知识去,就不重复记载了。
- voidAudioFlinger::PlaybackThread::threadLoop_standby()
- {
- ALOGV("Audiohardwareenteringstandby,mixer%p,suspendcount%d",this,mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
- }
- boolAudioFlinger::PlaybackThread::threadLoop()
- {
- //......
- while(!exitPending())
- <PREclass=cppname="code"><SPANstyle="FONT-FAMILY:Arial,Helvetica,sans-serif">{</SPAN></PRE>if(CC_UNLIKELY((!mActiveTracks.size()&&systemTime()>standbyTime)||isSuspended())){if(!mStandby){threadLoop_standby();mStandby=true;}//......}//......standbyTime=systemTime()+standbyDelay;//......}//......}
- voidAudioFlinger::PlaybackThread::threadLoop_standby()
- {
- ALOGV("Audiohardwareenteringstandby,mixer%p,suspendcount%d",this,mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
- }
- boolAudioFlinger::PlaybackThread::threadLoop()
- {
- //......
- while(!exitPending())
- <divclass="dp-highlighterbg_cpp"><divclass="bar"><divclass="tools"><strong>[cpp]</strong><atarget=_blankclass="ViewSource"title="viewplain"href="http://blog.csdn.net/special_lin/article/details/12849637#">viewplain</a><atarget=_blankclass="CopyToClipboard"title="copy"href="http://blog.csdn.net/special_lin/article/details/12849637#">copy</a><atarget=_blankclass="PrintSource"title="print"href="http://blog.csdn.net/special_lin/article/details/12849637#">print</a><atarget=_blankclass="About"title="?"href="http://blog.csdn.net/special_lin/article/details/12849637#">?</a></div></div><olclass="dp-cpp"><liclass="alt"><span><span><SPANstyle=</span><spanclass="string">"FONT-FAMILY:Arial,Helvetica,sans-serif"</span><span>>{</SPAN></span></span></li></ol></div><prestyle="DISPLAY:none"class="cpp"name="code"><spanstyle="font-family:Arial,Helvetica,sans-serif;"><span>{</span></span>
这里我们看到了,standby是由AudioFlinger控制的,一旦满足以下条件后,没有AudioTrack处于活动状态并且已经到达了standbyTime这个时间就进入Standby模式。那么standbyTime=systemTime() + standbyDelay,也就是过了standbyDelay这段时间后,音频系统将进入待机,关闭音频设备。最后找到standbyDelay的值是多少。
AudioFlinger::PlaybackThread构造函数中,将standbyDelay初始化,standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
AudioFlinger这个类第一次被引用时,就对成员变量mStandbyTimeInNsecs 进行了初始化
[cpp] view plain copy- voidAudioFlinger::onFirstRef()
- {
- //......
- /*TODO:moveallthisworkintoanInit()function*/
- charval_str[PROPERTY_VALUE_MAX]={0};
- if(property_get("ro.audio.flinger_standbytime_ms",val_str,NULL)>=0){
- uint32_tint_val;
- if(1==sscanf(val_str,"%u",&int_val)){
- mStandbyTimeInNsecs=milliseconds(int_val);
- ALOGI("Using%umSecasstandbytime.",int_val);
- }else{
- mStandbyTimeInNsecs=kDefaultStandbyTimeInNsecs;
- ALOGI("Usingdefault%umSecasstandbytime.",
- (uint32_t)(mStandbyTimeInNsecs/1000000));
- }
- }
- mMode=AUDIO_MODE_NORMAL;
- }
- voidAudioFlinger::onFirstRef()
- {
- //......
- /*TODO:moveallthisworkintoanInit()function*/
- charval_str[PROPERTY_VALUE_MAX]={0};
- if(property_get("ro.audio.flinger_standbytime_ms",val_str,NULL)>=0){
- uint32_tint_val;
- if(1==sscanf(val_str,"%u",&int_val)){
- mStandbyTimeInNsecs=milliseconds(int_val);
- ALOGI("Using%umSecasstandbytime.",int_val);
- }else{
- mStandbyTimeInNsecs=kDefaultStandbyTimeInNsecs;
- ALOGI("Usingdefault%umSecasstandbytime.",
- (uint32_t)(mStandbyTimeInNsecs/1000000));
- }
- }
- mMode=AUDIO_MODE_NORMAL;
- }
如果有ro.audio.flinger_standbytime_ms这个属性,就按这个属性值设定stand by的idle time(很可能是OEM代码),如果没有,取kDefaultStandbyTimeInNsecs的值。kDefaultStandbyTimeInNsecs是个常量,3s: [cpp] view plain copy
- staticconstnsecs_tkDefaultStandbyTimeInNsecs=seconds(3);
- staticconstnsecs_tkDefaultStandbyTimeInNsecs=seconds(3);
通过分析研究Android系统代码,我们虽然最终没有解决问题,但是已经定位出了问题所在的层次,确定这是一个驱动的BUG。Framework工程师的任务至此完成了。问题交付给驱动工程师,经过排查发现,是PA没有打开造成的问题。
经验可以带来技巧,如果下次遇到类似问题,我们可以直接在AudioHardware中截获PCM,通过判断解码出的PCM流是否正确,较快速的定位到问题所在——是MediaPlayer Codec、AudioSystem、还是Driver。
原文转自:http://blog.csdn.net/special_lin/article/details/12849637
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