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:
- protected void onCreate(Bundle savedInstanceState) {
- super.onCreate(savedInstanceState);
- setContentView(R.layout.activity_test_sound_pool);
- mSoundPool = new SoundPool(10, AudioManager.STREAM_SYSTEM, 5);
- mSoundId = mSoundPool.load(this, R.raw.camera_click, 1);
-
- vBtnShut = (Button) findViewById(R.id.btn_click);
- vBtnShut.setOnClickListener(new OnClickListener() {
- @Override
- public void onClick(View v) {
- mSoundPool.play(mSoundId, 1, 1, 0, 0, 1);
- }
- });
- }
-
- protected void onCreate(Bundle savedInstanceState) {
- super.onCreate(savedInstanceState);
- setContentView(R.layout.activity_test_sound_pool);
- mSoundPool = new SoundPool(10, AudioManager.STREAM_SYSTEM, 5);
- mSoundId = mSoundPool.load(this, R.raw.camera_click, 1);
-
- vBtnShut = (Button) findViewById(R.id.btn_click);
- vBtnShut.setOnClickListener(new OnClickListener() {
- @Override
- public void onClick(View v) {
- mSoundPool.play(mSoundId, 1, 1, 0, 0, 1);
- }
- });
- }
-
结果表明,BUG现象仍然是一样的。我们将BUG现象做一次简化:
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:
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。
- int SoundPool::load(int fd, int64_t offset, int64_t length, int priority)
- {
- ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
- fd, offset, length, priority);
- Mutex::Autolock lock(&mLock);
- sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length);
- mSamples.add(sample->sampleID(), sample);
- doLoad(sample);
- return sample->sampleID();
- }
- int SoundPool::load(int fd, int64_t offset, int64_t length, int priority)
- {
- ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
- fd, offset, length, priority);
- Mutex::Autolock lock(&mLock);
- sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length);
- mSamples.add(sample->sampleID(), sample);
- doLoad(sample);
- return sample->sampleID();
- }
数据处理角度来说,真正的load在doLoad中:
- void SoundPool::doLoad(sp<Sample>& sample)
- {
- ALOGV("doLoad: loading sample sampleID=%d", sample->sampleID());
- sample->startLoad();
- mDecodeThread->loadSample(sample->sampleID());
- }
- void SoundPool::doLoad(sp<Sample>& sample)
- {
- ALOGV("doLoad: loading sample sampleID=%d", sample->sampleID());
- sample->startLoad();
- mDecodeThread->loadSample(sample->sampleID());
- }
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:
- void SoundPoolThread::loadSample(int sampleID) {
- write(SoundPoolMsg(SoundPoolMsg::LOAD_SAMPLE, sampleID));
- }
- void SoundPoolThread::loadSample(int sampleID) {
- 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:
- int SoundPoolThread::run() {
- ALOGV("run");
- for (;;) {
- SoundPoolMsg msg = read();
- ALOGV("Got message m=%d, mData=%d", msg.mMessageType, msg.mData);
- switch (msg.mMessageType) {
- case SoundPoolMsg::KILL:
- ALOGV("goodbye");
- return NO_ERROR;
- case SoundPoolMsg::LOAD_SAMPLE:
- doLoadSample(msg.mData);
- break;
- default:
- ALOGW("run: Unrecognized message %d\n",
- msg.mMessageType);
- break;
- }
- }
- }
- int SoundPoolThread::run() {
- ALOGV("run");
- for (;;) {
- SoundPoolMsg msg = read();
- ALOGV("Got message m=%d, mData=%d", msg.mMessageType, msg.mData);
- switch (msg.mMessageType) {
- case SoundPoolMsg::KILL:
- ALOGV("goodbye");
- return NO_ERROR;
- case SoundPoolMsg::LOAD_SAMPLE:
- doLoadSample(msg.mData);
- break;
- default:
- ALOGW("run: Unrecognized message %d\n",
- msg.mMessageType);
- break;
- }
- }
- }
- void SoundPoolThread::doLoadSample(int sampleID) {
- sp <Sample> sample = mSoundPool->findSample(sampleID);
- status_t status = -1;
- if (sample != 0) {
- status = sample->doLoad();
- }
- mSoundPool->notify(SoundPoolEvent(SoundPoolEvent::SAMPLE_LOADED, sampleID, status));
- }
- void SoundPoolThread::doLoadSample(int sampleID) {
- sp <Sample> sample = mSoundPool->findSample(sampleID);
- status_t status = -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:
- status_t Sample::doLoad()
- {
- uint32_t sampleRate;
- int numChannels;
- audio_format_t format;
- sp<IMemory> p;
- ALOGV("Start decode");
- 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("Unable to load sample: %s", mUrl);
- return -1;
- }
- ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
- p->pointer(), p->size(), sampleRate, numChannels);
-
- if (sampleRate > kMaxSampleRate) {
- ALOGE("Sample rate (%u) out of range", sampleRate);
- return - 1;
- }
-
- if ((numChannels < 1) || (numChannels > 2)) {
- ALOGE("Sample channel count (%d) out of range", numChannels);
- return - 1;
- }
-
-
- uint8_t* q = static_cast<uint8_t*>(p->pointer()) + p->size() - 10;
-
-
- mData = p;
- mSize = p->size();
- mSampleRate = sampleRate;
- mNumChannels = numChannels;
- mFormat = format;
- mState = READY;
- return 0;
- }
- status_t Sample::doLoad()
- {
- uint32_t sampleRate;
- int numChannels;
- audio_format_t format;
- sp<IMemory> p;
- ALOGV("Start decode");
- 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("Unable to load sample: %s", mUrl);
- return -1;
- }
- ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
- p->pointer(), p->size(), sampleRate, numChannels);
-
- if (sampleRate > kMaxSampleRate) {
- ALOGE("Sample rate (%u) out of range", sampleRate);
- return - 1;
- }
-
- if ((numChannels < 1) || (numChannels > 2)) {
- ALOGE("Sample channel count (%d) out of range", numChannels);
- return - 1;
- }
-
-
- uint8_t* q = static_cast<uint8_t*>(p->pointer()) + p->size() - 10;
-
-
- mData = p;
- mSize = p->size();
- mSampleRate = sampleRate;
- mNumChannels = numChannels;
- mFormat = format;
- mState = READY;
- return 0;
- }
原来Sample请来了MediaPlayer帮其解码,并计算出了采样率和帧数。到这里数据已经准备好了。接下来我们就要看Framework能否把数据正确的传递给HAL,至于MediaPlayer是如何解码的我们先不研究。
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:
- int SoundPool::play(int sampleID, float leftVolume, float rightVolume,
- int priority, int loop, float rate)
- {
-
- channel = allocateChannel_l(priority);
-
- channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate);
-
- }
- int SoundPool::play(int sampleID, float leftVolume, float rightVolume,
- int priority, int loop, float rate)
- {
-
- channel = allocateChannel_l(priority);
-
- channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate);
-
- }
调用了SoundChannel的play.好读书而不求甚解,先把代码一路追下去,不作细究。
- void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume,
- float rightVolume, int priority, int loop, float rate)
- {
- AudioTrack* newTrack;
-
- newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
- channels, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData, bufferFrames);
-
- mState = PLAYING;
- mAudioTrack->start();
-
- }
- void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume,
- float rightVolume, int priority, int loop, float rate)
- {
- AudioTrack* newTrack;
-
- newTrack = new AudioTrack(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()。
- AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface*clientInterface)…
- {
-
- for (size_t i = 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_t i = 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:
- static audio_module_handle_t aps_load_hw_module(void*service,const char *name)
- {
- sp<IAudioFlinger> af= AudioSystem::get_audio_flinger();
- …
- return af->loadHwModule(name);
- }
- static audio_module_handle_t aps_load_hw_module(void*service,const char *name)
- {
- sp<IAudioFlinger> af= AudioSystem::get_audio_flinger();
- …
- return af->loadHwModule(name);
- }
AudioFlinger终于出现了,同样的情况也适用于mpClientInterface->openOutput,代码如下:
- static audio_io_handle_t aps_open_output(…)
- {
- sp<IAudioFlinger> af= AudioSystem::get_audio_flinger();
- …
- return af->openOutput((audio_module_handle_t)0,pDevices, pSamplingRate, pFormat, pChannelMask,
- pLatencyMs, flags);
- }
- static audio_io_handle_t aps_open_output(…)
- {
- sp<IAudioFlinger> af= AudioSystem::get_audio_flinger();
- …
- return af->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:
- 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)
- {
- #ifdef DEBUG_ALSA_OUT
- if(alsa_out_fp== NULL)
- alsa_out_fp = fopen("/data/data/out.pcm","a+");
- if(alsa_out_fp)
- ALOGI("------------>openfile success");
- #endif
- }
- ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t bytes)
- {
-
- #ifdef DEBUG_ALSA_OUT
- if(alsa_out_fp)
- fwrite(buffer,1,bytes,alsa_out_fp);
- #endif
-
- if (mStandby) {
- open_l();
- 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)
- {
- #ifdef DEBUG_ALSA_OUT
- if(alsa_out_fp== NULL)
- alsa_out_fp = fopen("/data/data/out.pcm","a+");
- if(alsa_out_fp)
- ALOGI("------------>openfile success");
- #endif
- }
- ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t bytes)
- {
-
- #ifdef DEBUG_ALSA_OUT
- if(alsa_out_fp)
- fwrite(buffer,1,bytes,alsa_out_fp);
- #endif
-
- if (mStandby) {
- open_l();
- 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():
- status_t AudioHardware::AudioStreamOutALSA::open_l()
- {
-
- mPcm = mHardware->openPcmOut_l();
- if (mPcm == NULL) {
- return NO_INIT;
- }
-
- }
- struct pcm *AudioHardware::openPcmOut_l()
- {
-
- mPcm = pcm_open(flags);
-
- if (!pcm_ready(mPcm)) {
- pcm_close(mPcm);
-
- }
- }
- return mPcm;
- }
- status_t AudioHardware::AudioStreamOutALSA::open_l()
- {
-
- mPcm = mHardware->openPcmOut_l();
- if (mPcm == NULL) {
- return NO_INIT;
- }
-
- }
- struct pcm *AudioHardware::openPcmOut_l()
- {
-
- mPcm = pcm_open(flags);
-
- if (!pcm_ready(mPcm)) {
- pcm_close(mPcm);
-
- }
- }
- return mPcm;
- }
open_l中调用了AudioHardware::openPcmOut_l,AudioHardware::openPcmOut_l中调用了pcm_open。再到pcm_open 中去看看:
- struct pcm *pcm_open(unsigned flags)
- {
-
-
- if (flags & PCM_IN) {
- dname = "/dev/snd/pcmC0D0c";
- channalFlags = -1;
- startCheckCount = 0;
- } else {
- #ifdef SUPPORT_USB
- dname = "/dev/snd/pcmC1D0p";
- #else
- dname = "/dev/snd/pcmC0D0p";
- #endif
- }
-
- pcm->fd = open(dname, O_RDWR);
- if (pcm->fd < 0) {
- oops(pcm, errno, "cannot open device '%s'", dname);
- return pcm;
- }
-
- if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_INFO, &info)) {
- oops(pcm, errno, "cannot get info - %s", dname);
- goto fail;
- }
-
- if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_HW_PARAMS, ¶ms)) {
- oops(pcm, errno, "cannot set hw params");
- goto fail;
- }
- if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_SW_PARAMS, &sparams)) {
- oops(pcm, errno, "cannot set sw params");
- goto fail;
- }
-
- fail:
- close(pcm->fd);
- pcm->fd = -1;
- return pcm;
- }
- struct pcm *pcm_open(unsigned flags)
- {
-
-
- if (flags & PCM_IN) {
- dname = "/dev/snd/pcmC0D0c";
- channalFlags = -1;
- startCheckCount = 0;
- } else {
- #ifdef SUPPORT_USB
- dname = "/dev/snd/pcmC1D0p";
- #else
- dname = "/dev/snd/pcmC0D0p";
- #endif
- }
-
- pcm->fd = open(dname, O_RDWR);
- if (pcm->fd < 0) {
- oops(pcm, errno, "cannot open device '%s'", dname);
- return pcm;
- }
-
- if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_INFO, &info)) {
- oops(pcm, errno, "cannot get info - %s", dname);
- goto fail;
- }
-
- if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_HW_PARAMS, ¶ms)) {
- oops(pcm, errno, "cannot set hw params");
- goto fail;
- }
- if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_SW_PARAMS, &sparams)) {
- oops(pcm, errno, "cannot set sw params");
- goto fail;
- }
-
- fail:
- close(pcm->fd);
- pcm->fd = -1;
- return pcm;
- }
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。
- <PRE class=cpp name="code"><PRE class=cpp name="code">status_t AudioHardware::AudioStreamOutALSA::standby()
- {
- doStandby_l();
- }
-
- void AudioHardware::AudioStreamOutALSA::doStandby_l()
- {
-
- if(!mStandby)
- mStandby = true;
- close_l();
- }
-
- void AudioHardware::AudioStreamOutALSA::close_l()
- {
- if (mPcm) {
- mHardware->closePcmOut_l();
- mPcm = NULL;
- }
- }</PRE><BR><BR></PRE>
- <div class="dp-highlighter bg_cpp"><div class="bar"><div class="tools"><strong>[cpp]</strong> <a target=_blank class="ViewSource" title="view plain" href="http://blog.csdn.net/special_lin/article/details/12849637#">view plain</a><a target=_blank class="CopyToClipboard" title="copy" href="http://blog.csdn.net/special_lin/article/details/12849637#">copy</a><a target=_blank class="PrintSource" title="print" href="http://blog.csdn.net/special_lin/article/details/12849637#">print</a><a target=_blank class="About" title="?" href="http://blog.csdn.net/special_lin/article/details/12849637#">?</a></div></div><ol class="dp-cpp"><li class="alt"><span><span><PRE </span><span class="keyword">class</span><span>=cpp name=</span><span class="string">"code"</span><span>>status_t AudioHardware::AudioStreamOutALSA::standby() </span></span></li><li><span>{ </span></li><li class="alt"><span> doStandby_l(); </span></li><li><span>} </span></li><li class="alt"><span> </span></li><li><span></span><span class="keyword">void</span><span> AudioHardware::AudioStreamOutALSA::doStandby_l() </span></li><li class="alt"><span>{ </span></li><li><span> </span></li><li class="alt"><span> </span><span class="keyword">if</span><span>(!mStandby) </span></li><li><span> mStandby = </span><span class="keyword">true</span><span>; </span></li><li class="alt"><span> close_l(); </span></li><li><span>} </span></li><li class="alt"><span> </span></li><li><span></span><span class="keyword">void</span><span> AudioHardware::AudioStreamOutALSA::close_l() </span></li><li class="alt"><span>{ </span></li><li><span> </span><span class="keyword">if</span><span> (mPcm) { </span></li><li class="alt"><span> mHardware->closePcmOut_l(); </span></li><li><span> mPcm = NULL; </span></li><li class="alt"><span> } </span></li><li><span>}</PRE><BR><BR> </span></li></ol></div><pre style="DISPLAY: none" class="cpp" name="code"><div class="dp-highlighter bg_cpp"><div class="bar"><div class="tools"><strong>[cpp]</strong> <a target=_blank class="ViewSource" title="view plain" href="http://blog.csdn.net/special_lin/article/details/12849637#">view plain</a><a target=_blank class="CopyToClipboard" title="copy" href="http://blog.csdn.net/special_lin/article/details/12849637#">copy</a><a target=_blank class="PrintSource" title="print" href="http://blog.csdn.net/special_lin/article/details/12849637#">print</a><a target=_blank class="About" title="?" href="http://blog.csdn.net/special_lin/article/details/12849637#">?</a></div></div><ol class="dp-cpp"><li class="alt"><span><span>status_t AudioHardware::AudioStreamOutALSA::standby() </span></span></li><li><span>{ </span></li><li class="alt"><span> doStandby_l(); </span></li><li><span>} </span></li><li class="alt"><span> </span></li><li><span></span><span class="keyword">void</span><span> AudioHardware::AudioStreamOutALSA::doStandby_l() </span></li><li class="alt"><span>{ </span></li><li><span> </span></li><li class="alt"><span> </span><span class="keyword">if</span><span>(!mStandby) </span></li><li><span> mStandby = </span><span class="keyword">true</span><span>; </span></li><li class="alt"><span> close_l(); </span></li><li><span>} </span></li><li class="alt"><span> </span></li><li><span></span><span class="keyword">void</span><span> AudioHardware::AudioStreamOutALSA::close_l() </span></li><li class="alt"><span>{ </span></li><li><span> </span><span class="keyword">if</span><span> (mPcm) { </span></li><li class="alt"><span> mHardware->closePcmOut_l(); </span></li><li><span> mPcm = NULL; </span></li><li class="alt"><span> } </span></li><li><span>} </span></li></ol></div><pre style="DISPLAY: none" class="cpp" name="code">status_t AudioHardware::AudioStreamOutALSA::standby()
- {
- doStandby_l();
- }
-
- void AudioHardware::AudioStreamOutALSA::doStandby_l()
- {
-
- if(!mStandby)
- mStandby = true;
- close_l();
- }
-
- void AudioHardware::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。
- void AudioFlinger::PlaybackThread::threadLoop_standby()
- {
- ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
- }
-
- bool AudioFlinger::PlaybackThread::threadLoop()
- {
-
- while (!exitPending())
- <PRE class=cpp name="code"> <SPAN style="FONT-FAMILY: Arial, Helvetica, sans-serif">{</SPAN></PRE> if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || isSuspended())) { if (!mStandby) { threadLoop_standby(); mStandby = true; }
- void AudioFlinger::PlaybackThread::threadLoop_standby()
- {
- ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
- }
-
- bool AudioFlinger::PlaybackThread::threadLoop()
- {
-
- while (!exitPending())
- <div class="dp-highlighter bg_cpp"><div class="bar"><div class="tools"><strong>[cpp]</strong> <a target=_blank class="ViewSource" title="view plain" href="http://blog.csdn.net/special_lin/article/details/12849637#">view plain</a><a target=_blank class="CopyToClipboard" title="copy" href="http://blog.csdn.net/special_lin/article/details/12849637#">copy</a><a target=_blank class="PrintSource" title="print" href="http://blog.csdn.net/special_lin/article/details/12849637#">print</a><a target=_blank class="About" title="?" href="http://blog.csdn.net/special_lin/article/details/12849637#">?</a></div></div><ol class="dp-cpp"><li class="alt"><span><span><SPAN style=</span><span class="string">"FONT-FAMILY: Arial, Helvetica, sans-serif"</span><span>>{</SPAN> </span></span></li></ol></div><pre style="DISPLAY: none" class="cpp" name="code"> <span style="font-family:Arial, Helvetica, sans-serif;"><span>{</span></span>
if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || isSuspended())) { if (!mStandby) { threadLoop_standby(); mStandby = true; }//... ...}//... ...standbyTime = systemTime() + standbyDelay;//... ...}// ... ...}
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
- void AudioFlinger::onFirstRef()
- {
-
-
- 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));
- }
- }
-
- mMode = AUDIO_MODE_NORMAL;
- }
- void AudioFlinger::onFirstRef()
- {
-
-
- 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));
- }
- }
-
- 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:
- static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
- static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
五. 结论及收获
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