Andorid QQ for TV麦克调试问题

1.Android系统默认采样率是8000Hz；

2.QQ应用采样率是44100Hz或16000Hz；

3.QQ for TV应用会在真正开始语音聊天前，测试最佳采样率(即先会执行打开和关闭设备的一个测试；注意：有些驱动对快速打开关闭会有异常；这时可以考虑在HAL设置时间戳，当调用HAL关闭和打开函数的时间间隔不超过1S时、不执行关闭动作；完后记录状态，再次打开时也不再执行打开动作)；

4.重采样由AudioFlinger服务完成，不需要驱动和HAL模块参与。

一、流程

E/AudioRecord( 2019): Could not get audio input for record source 1

E/AudioRecord-JNI( 2019): Error creating AudioRecord instance: initialization check failed.

E/AudioRecord-Java( 2019): [ android.media.AudioRecord ] Error code -20 when initializing native AudioRecord object.

如上错误是当前mic的配置文件中不存在应用设置的采样率、采样精度或声道。

frameworks/base/media/java/android/media/AudioRecord.java

public AudioRecord(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat,             int bufferSizeInBytes)    throws IllegalArgumentException {  int initResult = native_setup( new WeakReference<AudioRecord>(this),                 mRecordSource, mSampleRate, mChannels, mAudioFormat, mNativeBufferSizeInBytes,                session);  if (initResult != SUCCESS) {            loge("Error code "+initResult+" when initializing native AudioRecord object.");            return; // with mState == STATE_UNINITIALIZED  }}

frameworks/base/core/jni/android_media_AudioRecord.cpp

{"native_setup",         "(Ljava/lang/Object;IIIII[I)I",                                       (void *)android_media_AudioRecord_setup},static intandroid_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,        jint source, jint sampleRateInHertz, jint channels,        jint audioFormat, jint buffSizeInBytes, jintArray jSession){  sp<AudioRecord> lpRecorder = new AudioRecord();  lpRecorder->set((audio_source_t) source,        sampleRateInHertz,        format,        // word length, PCM        channels,        frameCount,        recorderCallback,// callback_t        lpCallbackData,// void* user        0,             // notificationFrames,        true,          // threadCanCallJava)        sessionId);  if (lpRecorder->initCheck() != NO_ERROR) {    ALOGE("Error creating AudioRecord instance: initialization check failed.");    goto native_init_failure;  }  }

frameworks/av/media/libmedia/AudioRecord.cpp

status_t AudioRecord::set(        audio_source_t inputSource,        uint32_t sampleRate,        audio_format_t format,        audio_channel_mask_t channelMask,        int frameCount,        callback_t cbf,        void* user,        int notificationFrames,        bool threadCanCallJava,        int sessionId){  audio_io_handle_t input = AudioSystem::getInput(inputSource,                                                    sampleRate,                                                    format,                                                    channelMask,                                                    mSessionId);  if (input == 0) {    ALOGE("Could not get audio input for record source %d", inputSource);    return BAD_VALUE;  }}audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,                                    uint32_t samplingRate,                                    audio_format_t format,                                    audio_channel_mask_t channelMask,                                    int sessionId){  const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();  if (aps == 0) return 0;  return aps->getInput(inputSource, samplingRate, format, channelMask, sessionId);}

Binder进程间通信

frameworks/av/services/audioflinger/AudioPolicyService.cpp

audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,                                    uint32_t samplingRate,                                    audio_format_t format,                                    audio_channel_mask_t channelMask,                                    int audioSession){  audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,                                                       format, channelMask, (audio_in_acoustics_t) 0);  //rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);  //rc = audio_policy_dev_open(module, &mpAudioPolicyDev);}

hardware/libhardware_legacy/audio/Audio_policy.c

static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,                                      uint32_t sampling_rate,                                      audio_format_t format,                                      audio_channel_mask_t channelMask,                                      audio_in_acoustics_t acoustics){  struct legacy_audio_policy *lap = to_lap(pol);  return lap->apm->getInput((int) inputSource, sampling_rate, (int) format, channelMask,                              (AudioSystem::audio_in_acoustics)acoustics);}

hardware/libhardware_legacy/audio/AudiopolicyManagerBase.cpp

audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,                                    uint32_t samplingRate,                                    uint32_t format,                                    uint32_t channelMask,                                    AudioSystem::audio_in_acoustics acoustics){  audio_devices_t device = getDeviceForInputSource(inputSource);  //查询当前麦克  IOProfile *profile = getInputProfile(device,                                         samplingRate,                                         format,                                         channelMask);  //根据audio_policy.conf中记录当前麦克信息，看是否支持采样率、采样位数以及声道。  if (profile == NULL) {    ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d,"                "channelMask %04x",                device, samplingRate, format, channelMask);    return 0;  }  input = mpClientInterface->openInput(profile->mModule->mHandle,                                    &inputDesc->mDevice,                                    &inputDesc->mSamplingRate,                                    &inputDesc->mFormat,                                    &inputDesc->mChannelMask);}

frameworks/av/services/audioflinger/AudioPolicyService.cpp

open_input_on_module  : aps_open_input_on_module,static audio_io_handle_t aps_open_input_on_module(void *service,                                                  audio_module_handle_t module,                                                  audio_devices_t *pDevices,                                                  uint32_t *pSamplingRate,                                                  audio_format_t *pFormat,                                                  audio_channel_mask_t *pChannelMask){  sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();  if (af == 0) {    ALOGW("%s: could not get AudioFlinger", __func__);    return 0;  }  return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);}

frameworks/av/services/audioflinger/AudioFlinger.cpp

audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,                                          audio_devices_t *pDevices,                                          uint32_t *pSamplingRate,                                          audio_format_t *pFormat,                                          audio_channel_mask_t *pChannelMask){  status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,                                        &inStream);status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,                                        &inStream);  if (status == NO_ERROR && inStream != NULL) {    thread = new RecordThread(this,                                  input,                                  reqSamplingRate,                                  reqChannels,                                  id,                                  device);  }}AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,                                         AudioStreamIn *input,                                         uint32_t sampleRate,                                         audio_channel_mask_t channelMask,                                         audio_io_handle_t id,                                         audio_devices_t device) :    ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD),    mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),    // mRsmpInIndex and mInputBytes set by readInputParameters()    mReqChannelCount(popcount(channelMask)),    mReqSampleRate(sampleRate)    // mBytesRead is only meaningful while active, and so is cleared in start()    // (but might be better to also clear here for dump?){  readInputParameters()}void AudioFlinger::RecordThread::readInputParameters(){  mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);  if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2){    mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);    mResampler->setSampleRate(mSampleRate);  }  //重采样！！！}bool AudioFlinger::RecordThread::threadLoop(){  if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {    //正常从HAL获取数据  }  else{    nsecs_t now = systemTime();    if ((now - lastWarning) > kWarningThrottleNs) {      ALOGW("RecordThread: buffer overflow");      lastWarning = now;    }  }}

二、应用如何拿取buffer数据

W/AudioFlinger(  918): RecordThread: buffer overflow

W/AudioRecord( 2238): obtainBuffer timed out (is the CPU pegged?) user=0002f260, server=0002f260

应该考虑buffer溢出；尤其是第二个问题，应该考虑底层音频数据有问题。

frameworks/base/media/java/android/media/AudioRecord.java

public int read(byte[] audioData, int offsetInBytes, int sizeInBytes) {  if (mState != STATE_INITIALIZED) {    return ERROR_INVALID_OPERATION;  }        if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)                 || (offsetInBytes + sizeInBytes > audioData.length)) {    return ERROR_BAD_VALUE;  }  return native_read_in_byte_array(audioData, offsetInBytes, sizeInBytes);}

frameworks/base/core/jni/android_media_AudioRecord.cpp

static jint android_media_AudioRecord_readInByteArray(JNIEnv *env,  jobject thiz,                                                        jbyteArray javaAudioData,                                                        jint offsetInBytes, jint sizeInBytes) {  sp<AudioRecord> lpRecorder = getAudioRecord(env, thiz);  ssize_t readSize = lpRecorder->read(recordBuff + offsetInBytes,                                        sizeInBytes > (jint)recorderBuffSize ?                                            (jint)recorderBuffSize : sizeInBytes );}

frameworks/av/media/libmedia/AudioRecord.cpp

ssize_t AudioRecord::read(void* buffer, size_t userSize){  status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS));}status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount){  cblk->waitTimeMs += waitTimeMs;  if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {    ALOGW(   "obtainBuffer timed out (is the CPU pegged?) "                            "user=%08x, server=%08x", cblk->user, cblk->server);  }  audioBuffer->raw         = (int8_t*)cblk->buffer(u);  //cblk->buffer(u)即是AudioFlinger中mActiveTrack->getNextBuffer(&buffer)读取的HAL数据}