问题：
当不插usb声卡时，打开录像机，点录像键，几秒钟后系统提示错误，退出，再次点击录像键或照相机，均打不开
用录像机录像时，要打开音频输入设备
在AudioFlinger层
frameworks/base/services/audioflinger/AudioFlinger.cpp
int AudioFlinger::openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t acoustics)
{
status_t status;
RecordThread *thread = NULL;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
uint32_t format = pFormat ? *pFormat : 0;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t reqSamplingRate = samplingRate;
uint32_t reqFormat = format;
uint32_t reqChannels = channels;
audio_stream_in_t *inStream;
audio_hw_device_t *inHwDev;
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
inHwDev = findSuitableHwDev_l(*pDevices);
if (inHwDev == NULL)
return 0;
status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
&channels, &samplingRate,
(audio_in_acoustics_t)acoustics,
&inStream); //问题出在这，当没有USB声卡时，这个函数没有返回。
LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
inStream,
samplingRate,
format,
channels,
acoustics,
status);
// If the input could not be opened with the requested parameters and we can handle the conversion internally,
// try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
// or stereo to mono conversions on 16 bit PCM inputs.
if (inStream == NULL && status == BAD_VALUE &&
reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
(samplingRate <= 2 * reqSamplingRate) &&
(getInputChannelCount(channels) < 3) && (getInputChannelCount(reqChannels) < 3)) {
LOGV("openInput() reopening with proposed sampling rate and channels");
status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
&channels, &samplingRate,
(audio_in_acoustics_t)acoustics,
&inStream);
}
if (inStream != NULL) {
AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
int id = nextUniqueId();
// Start record thread
// RecorThread require both input and output device indication to forward to audio
// pre processing modules
uint32_t device = (*pDevices) | primaryOutputDevice_l();
thread = new RecordThread(this,
input,
reqSamplingRate,
reqChannels,
id,
device);
mRecordThreads.add(id, thread);
LOGV("openInput() created record thread: ID %d thread %p", id, thread);
if (pSamplingRate) *pSamplingRate = reqSamplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = reqChannels;
input->stream->common.standby(&input->stream->common);
// notify client processes of the new input creation
thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
return id;
}
return 0;
}
在AudioHardware层
hardware/qcom/media/audio/msm8660/audio_hw_hal.cpp
static int qcom_adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct qcom_audio_device *qadev;
int ret;
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
qadev = (struct qcom_audio_device *)calloc(1, sizeof(*qadev));
if (!qadev)
return -ENOMEM;
qadev->device.common.tag = HARDWARE_DEVICE_TAG;
qadev->device.common.version = 0;
qadev->device.common.module = const_cast<hw_module_t*>(module);
qadev->device.common.close = qcom_adev_close;
qadev->device.get_supported_devices = adev_get_supported_devices;
qadev->device.init_check = adev_init_check;
qadev->device.set_voice_volume = adev_set_voice_volume;
qadev->device.set_master_volume = adev_set_master_volume;
qadev->device.set_fm_volume = adev_set_fm_volume;
qadev->device.set_mode = adev_set_mode;
qadev->device.set_mic_mute = adev_set_mic_mute;
qadev->device.get_mic_mute = adev_get_mic_mute;
qadev->device.set_parameters = adev_set_parameters;
qadev->device.get_parameters = adev_get_parameters;
qadev->device.get_input_buffer_size = adev_get_input_buffer_size;
qadev->device.open_output_stream = adev_open_output_stream;
qadev->device.open_output_session = adev_open_output_session;
qadev->device.close_output_stream = adev_close_output_stream;
qadev->device.open_input_stream = adev_open_input_stream; //指向adev_open_input_stream（）
qadev->device.close_input_stream = adev_close_input_stream;
qadev->device.dump = adev_dump;
qadev->hwif = createAudioHardware();
if (!qadev->hwif) {
ret = -EIO;
goto err_create_audio_hw;
}
*device = &qadev->device.common;
return 0;
err_create_audio_hw:
free(qadev);
return ret;
}
/** This method creates and opens the audio hardware input stream */
static int adev_open_input_stream(struct audio_hw_device *dev,
uint32_t devices, int *format,
uint32_t *channels, uint32_t *sample_rate,
audio_in_acoustics_t acoustics,
struct audio_stream_in **stream_in)
{
struct legacy_audio_device *ladev = to_ladev(dev);
status_t status;
struct legacy_stream_in *in;
int ret;
in = (struct legacy_stream_in *)calloc(1, sizeof(*in));
if (!in)
return -ENOMEM;
in->legacy_in = ladev->hwif->openInputStream(devices, format, channels,
sample_rate, &status,
(AudioSystem::audio_in_acoustics)acoustics); //这里没有返回
if (!in->legacy_in) {
ret = status;
goto err_open;
}
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
in->stream.common.get_buffer_size = in_get_buffer_size;
in->stream.common.get_channels = in_get_channels;
in->stream.common.get_format = in_get_format;
in->stream.common.set_format = in_set_format;
in->stream.common.standby = in_standby;
in->stream.common.dump = in_dump;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
in->stream.common.add_audio_effect = in_add_audio_effect;
in->stream.common.remove_audio_effect = in_remove_audio_effect;
in->stream.set_gain = in_set_gain;
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
*stream_in = &in->stream;
return 0;
err_open:
free(in);
*stream_in = NULL;
return ret;
}
hardware/qcom/media/audio/msm8660/AudioHardware.cpp
887 AudioStreamIn* AudioHardware::openInputStream(
888 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
889 AudioSystem::audio_in_acoustics acoustic_flags)
890 {
891 // check for valid input source
892 if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
893 return 0;
894 }
895
896 #if 1 //WNC:WJ Wang 20110719: Add USB-Mic path: BEGIN
897 if (devices == AudioSystem::DEVICE_IN_USB_MIC)
898 {
899 Mutex::Autolock lock(mLock);
900
901 AudioStreamInUSBMic *in_usb = new AudioStreamInUSBMic();
902 status_t lStatus = in_usb->set(this, devices, format, channels, sampleRate, acoustic_flags);
903
904 if (status) {
906 *status = lStatus;
907 }
908 if (lStatus != NO_ERROR) //If open USB-Mic failed, continue to open the analog Mic
909 {
910 LOGE("before delete in_usb\n");
911 delete in_usb; //这句之后的log没有打出来。delete in_usb对象时，进入AudioStreamInUSBMic的析构函数
912 LOGE("delete in_usb\n");
913 }
914 else
915 {
916 mUsbInputs.add(in_usb);
917 return in_usb;
918 }
919 }
920 #endif //WNC:WJ Wang 20110719: Add USB-Mic path: END
922 mLock.lock();
923 if(devices == AudioSystem::DEVICE_IN_COMMUNICATION) {
924 LOGE("Create Audio stream Voip \n");
925 AudioStreamInVoip* inVoip = new AudioStreamInVoip();
926 status_t lStatus = NO_ERROR;
927 lStatus = inVoip->set(this, devices, format, channels, sampleRate, acoustic_flags);
928 if (status) {
929 *status = lStatus;
930 }
931 if (lStatus != NO_ERROR) {
932 LOGE(" Error creating voip input \n");
933 mLock.unlock();
934 delete inVoip;
935 return 0;
936 }
937 mVoipInputs.add(inVoip);
938 mLock.unlock();
939 return inVoip;
940 } else {
941 AudioStreamInMSM72xx* in72xx = new AudioStreamInMSM72xx();
942 status_t lStatus = in72xx->set(this, devices, format, channels, sampleRate, acoustic_flags);
943 if (status) {
944 *status = lStatus;
945 }
946 if (lStatus != NO_ERROR) {
947 LOGE("Error creating Audio stream AudioStreamInMSM72xx \n");
948 mLock.unlock();
949 delete in72xx;
950 return 0;
951 }
952 mInputs.add(in72xx);
953 mLock.unlock();
954 return in72xx;
955 }
956 }AudioHardware::AudioStreamInUSBMic::~AudioStreamInUSBMic() { standby(); }status_t AudioHardware::AudioStreamInUSBMic::standby() { AudioHardware *hw = mHardware; if (!mHardware) return -1; hw->mLock.lock(); if (mState > AUDIO_INPUT_CLOSED) { if (mHandle != 0) { snd_pcm_close(mHandle); mHandle = 0; if (mHardware != 0) { if(mHardware->mNumPcmRec > 0 && mFormat == AUDIO_HW_IN_FORMAT) mHardware->mNumPcmRec--; } } mState = AUDIO_INPUT_CLOSED; } hw->mLock.unlock(); return NO_ERROR; }
找到问题了：
在执行delete in_usb之前，执行了Mutex::Autolock lock(mLock); mLock被锁住了，所以在析构函数所调用的standby函数里，调用hw->mLock.lock();时，就获取不到锁，导致死锁！
AutoLock类是定义在Mutex内部的一个类，AutoLock的定义代码如下所示：
[-->Thread.h Mutex::Autolock声明和实现]
class Autolock { public: //构造的时候调用lock。 inline Autolock(Mutex& mutex) : mLock(mutex) { mLock.lock(); } inline Autolock(Mutex* mutex) : mLock(*mutex) { mLock.lock(); } //析构的时候调用unlock。 inline ~Autolock() { mLock.unlock(); } private: Mutex& mLock; };
AutoLock的用法很简单： 先定义一个Mutex，如 Mutex xlock。 在使用xlock的地方，定义一个AutoLock，如 AutoLock autoLock（xlock）。由于C++对象的构造和析构函数都是自动被调用的，所以在AutoLock的生命周期内，xlock的lock和unlock也就自动被调用了，这样就省去了重复书写unlock的麻烦，而且lock和unlock的调用肯定是一一对应的，这样就绝对不会出错。
在作用域里加锁，脱离作用域就会自动解锁。作用域是指同级{}范围。
Mutex::Autolock lock(mLock)的mLock在class AudioHardware中定义，所以和hw->mLock.lock();中的mLock是同一个锁。
相关代码：
set函数由openInputStream（）调用，没有usb声卡时返回BAD_VALUE，否则返回NO_ERROR。
status_t AudioHardware::AudioStreamInUSBMic::set( AudioHardware* hw, uint32_t devices, int *pFormat, uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustic_flags) { char cap_pcm[16]; LOGD("AudioHardware::AudioStreamInUSBMic::set(%d)...mState=%d", *pFormat, *pChannels); if (pFormat == 0 || pRate == 0 || pChannels == 0) { return BAD_VALUE; } uint32_t rate = hw->getInputSampleRate(*pRate); if (rate != *pRate) { *pRate = rate; return BAD_VALUE; } //Eaddy temp if (rate != 16000) rate = 16000; if ((*pChannels & (AudioSystem::CHANNEL_IN_MONO | AudioSystem::CHANNEL_IN_STEREO)) == 0) { *pChannels = AUDIO_HW_IN_CHANNELS; return BAD_VALUE; } if (mHandle != 0) { LOGE("Audio record already set"); return NO_ERROR; } mHardware = hw; if(mHardware->mNumPcmRec > 0) { /* Only one PCM recording is allowed at a time */ LOGE("Multiple PCM recordings is not allowed"); return -1; } status_t status = NO_ERROR; int dir = 0; snd_pcm_t *handle = 0; snd_pcm_hw_params_t *params = 0; status = getCaptureDevice(cap_pcm); if (status < 0) { LOGE("No USB-Audio:"); return BAD_VALUE; } /* Open PCM device for recording (capture). Old setting == "hw:2,0,0" */ status = snd_pcm_open(&handle, cap_pcm, SND_PCM_STREAM_CAPTURE, 0); if (status < 0) { LOGE("unable to open pcm device"); return status; } mHandle = handle; mHardware->mNumPcmRec++; mDevices = devices; mFormat = AUDIO_HW_IN_FORMAT; mChannels = *pChannels; mChannelCount = AudioSystem::popCount(mChannels); mSampleRate = rate; mAcoustics = acoustic_flags; mBufferSize = mHardware->getInputBufferSize(mSampleRate, AUDIO_HW_IN_FORMAT, mChannelCount); mBufferSize <<= 1; // We double the size of input buffer for ping pong use of record buffer. LOGD("AudioStreamInUSBMic::set(%d)...(%d, %d, %d)", __LINE__, mChannelCount, mSampleRate, mBufferSize); /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(¶ms); if (params == 0) { LOGE("unable to allocate parameter memory"); goto Error; } /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, mChannelCount); snd_pcm_uframes_t lvFrames; /* Set period size to frames. */ lvFrames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &lvFrames, &dir); /* Write the parameters to the driver */ status = snd_pcm_hw_params(handle, params); if (status < 0) { LOGE("unable to set hw params"); goto Error; } mState = AUDIO_INPUT_OPENED; return NO_ERROR; Error: if (mHandle != 0) { snd_pcm_close(mHandle); mHandle = 0; } return -1; }
getCaptureDevice，找到USB-Audio时，返回0，找不到时返回-1
int AudioHardware::AudioStreamInUSBMic::getCaptureDevice(char * cap_pcm_str) { int rc,idx,dev; snd_ctl_t *handle; snd_pcm_hw_params_t *params; snd_ctl_card_info_t *info; snd_pcm_info_t *pcminfo; char str[16]; char usb_card_str[16]; int cap_pcm_dev; int cap_pcm_sub_dev; snd_ctl_card_info_alloca(&info); snd_pcm_info_alloca(&pcminfo); idx = -1; while (1) { if ((rc = snd_card_next(&idx)) < 0) { LOGE("Card next error "); break; } if (idx < 0) break; /* Check sound card */ sprintf(str, "hw:CARD=%i", idx); if ((rc = snd_ctl_open(&handle, str, 0)) < 0) { LOGE("Open error: card = %s\n", str); continue; } if ((rc = snd_ctl_card_info(handle, info)) < 0) { LOGE("HW info error:"); continue; } /* Find "USB-Audio" */ sprintf(usb_card_str, "%s", snd_ctl_card_info_get_driver(info)); LOGI("Soundcard Driver = %s", usb_card_str); if ((rc = strncmp(usb_card_str, "USB-Audio", 9)) == 0) { LOGI("[--------------Got USB-Audio--------------]"); dev = -1; while (1) { snd_pcm_sync_id_t sync; if ((rc = snd_ctl_pcm_next_device(handle, &dev)) < 0) { LOGE(" PCM next device error "); break; } if (dev < 0) break; snd_pcm_info_set_device(pcminfo, dev); snd_pcm_info_set_subdevice(pcminfo, 0); snd_pcm_info_set_stream(pcminfo, SND_PCM_STREAM_CAPTURE); if ((rc = snd_ctl_pcm_info(handle, pcminfo)) < 0) { LOGE(" PCM info error"); continue; } /* Find the PCM capture device */ cap_pcm_dev = snd_pcm_info_get_device(pcminfo); cap_pcm_sub_dev = snd_pcm_info_get_subdevice(pcminfo); sprintf(cap_pcm_str, "hw:%i,%i,%i", idx, cap_pcm_dev, cap_pcm_sub_dev); LOGD("Open capture pcm device [%s]\n", cap_pcm_str); return 0; } }else{ continue; } } return idx; }
class AudioHardware定义：hardware/qcom/media/audio/msm8660/AudioHardware.h
class AudioHardware : public AudioHardwareBase { class AudioStreamOutMSM72xx; class AudioStreamInMSM72xx; public: AudioHardware(); virtual ~AudioHardware(); virtual status_t initCheck(); virtual status_t setVoiceVolume(float volume); virtual status_t setMasterVolume(float volume); virtual status_t setMode(int mode); // mic mute virtual status_t setMicMute(bool state); virtual status_t getMicMute(bool* state); virtual status_t setParameters(const String8& keyValuePairs); virtual String8 getParameters(const String8& keys); // create I/O streams virtual AudioStreamOut* openOutputStream( uint32_t devices, int *format=0, uint32_t *channels=0, uint32_t *sampleRate=0, status_t *status=0); virtual AudioStreamIn* openInputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics); virtual void closeOutputStream(AudioStreamOut* out); virtual void closeInputStream(AudioStreamIn* in); virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); void clearCurDevice() { mCurSndDevice = -1; } protected: virtual status_t dump(int fd, const Vector<String16>& args); private: status_t doAudioRouteOrMute(uint32_t device); status_t setMicMute_nosync(bool state); status_t checkMicMute(); status_t dumpInternals(int fd, const Vector<String16>& args); uint32_t getInputSampleRate(uint32_t sampleRate); bool checkOutputStandby(); status_t doRouting(AudioStreamInMSM72xx *input); AudioStreamInMSM72xx* getActiveInput_l(); class AudioStreamOutMSM72xx : public AudioStreamOut { public: AudioStreamOutMSM72xx(); virtual ~AudioStreamOutMSM72xx(); status_t set(AudioHardware* mHardware, uint32_t devices, int *pFormat, uint32_t *pChannels, uint32_t *pRate); virtual uint32_t sampleRate() const { return 44100; } // must be 32-bit aligned - driver only seems to like 4800 virtual size_t bufferSize() const { return 4800; } virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; } virtual int format() const { return AudioSystem::PCM_16_BIT; } virtual uint32_t latency() const { return (1000*AUDIO_HW_NUM_OUT_BUF*(bufferSize()/frameSize()))/sampleRate()+AUDIO_HW_OUT_LATENCY_MS; } virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; } virtual ssize_t write(const void* buffer, size_t bytes); virtual status_t standby(); virtual status_t dump(int fd, const Vector<String16>& args); bool checkStandby(); virtual status_t setParameters(const String8& keyValuePairs); virtual String8 getParameters(const String8& keys); uint32_t devices() { return mDevices; } virtual status_t getRenderPosition(uint32_t *dspFrames); private: AudioHardware* mHardware; int mFd; int mStartCount; int mRetryCount; bool mStandby; uint32_t mDevices; }; class AudioStreamInMSM72xx : public AudioStreamIn { public: enum input_state { AUDIO_INPUT_CLOSED, AUDIO_INPUT_OPENED, AUDIO_INPUT_STARTED }; AudioStreamInMSM72xx(); virtual ~AudioStreamInMSM72xx(); status_t set(AudioHardware* mHardware, uint32_t devices, int *pFormat, uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics); virtual size_t bufferSize() const { return mBufferSize; } virtual uint32_t channels() const { return mChannels; } virtual int format() const { return mFormat; } virtual uint32_t sampleRate() const { return mSampleRate; } virtual status_t setGain(float gain) { return INVALID_OPERATION; } virtual ssize_t read(void* buffer, ssize_t bytes); virtual status_t dump(int fd, const Vector<String16>& args); virtual status_t standby(); virtual status_t setParameters(const String8& keyValuePairs); virtual String8 getParameters(const String8& keys); virtual unsigned int getInputFramesLost() const { return 0; } uint32_t devices() { return mDevices; } int state() const { return mState; } virtual status_t addAudioEffect(effect_interface_s**) { return 0;} virtual status_t removeAudioEffect(effect_interface_s**) { return 0;} private: AudioHardware* mHardware; int mFd; int mState; int mRetryCount; int mFormat; uint32_t mChannels; uint32_t mSampleRate; size_t mBufferSize; AudioSystem::audio_in_acoustics mAcoustics; uint32_t mDevices; bool mFirstread; }; static const uint32_t inputSamplingRates[]; bool mInit; bool mMicMute; bool mBluetoothNrec; uint32_t mBluetoothId; AudioStreamOutMSM72xx* mOutput; SortedVector <AudioStreamInMSM72xx*> mInputs; msm_snd_endpoint *mSndEndpoints; int mNumSndEndpoints; int mCurSndDevice; int m7xsnddriverfd; bool mDualMicEnabled; int mTtyMode; friend class AudioStreamInMSM72xx; Mutex mLock; };
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