项目场景：

基于 owt 的 webrtc 应用，windows exe 和 chrome 浏览器建立连接，做一个类似远程控制的工具（例如向日葵）

问题描述：

windows 应用和 chrome 建立连接后听不到声音。声音是通过业务层参数传下来推流(publish) 或者拉流 (subscribe) 的

原因分析：

owt webrtc 线程调度关系，有一个初始化线程，还有一个取数据线程，取数据的线程发现取数据异常，就退出了，导致一个音频包都没有发

 //=======打印显示音频采集打开了=========
(core_audio_input_win.cc:276): --- Input audio stream is alive ---
(audio_device_buffer.cc:239): Size of recording buffer: 960
(render_delay_buffer.cc:362): Applying total delay of 5 blocks.
(matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms.
(matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms.
(matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms.
(matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms.
(matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms.
(render_delay_buffer.cc:330): Receiving a first externally reported audio buffer delay of 16 ms.
// ==========发了第一个音频包=============
(rtp_sender_audio.cc:309): First audio RTP packet sent to pacer
// ==========发了第一个视频包=============
(rtp_sender_video.cc:667): Sent first RTP packet of the first video frame (pre-pacer)
(rtp_sender_video.cc:671): Sent last RTP packet of the first video frame (pre-pacer)
(video_send_stream_impl.cc:467): SignalEncoderActive, Encoder is active.

异常的打印：（没有发第一个音频包的逻辑）

//========windows 音频 api 打开报错===========
(core_audio_base_win.cc:955): [Input] WASAPI streaming failed.
(channel.cc:821): Changing voice state, recv=0 send=1
(thread.cc:668): Message took 307ms to dispatch. Posted from: cricket::BaseChannel::UpdateMediaSendRecvState@../../third_party/webrtc/pc/channel.cc:800
(webrtc_video_engine.cc:1193): SetSend: true
(video_send_stream.cc:148): UpdateActiveSimulcastLayers: {1}
(bitrate_allocator.cc:523): UpdateAllocationLimits : total_requested_min_bitrate: 62 kbps, total_requested_padding_bitrate: 0 bps, total_requested_max_bitrate: 2532 kbps
(pacing_controller.cc:213): bwe:pacer_updated pacing_kbps=750 padding_budget_kbps=0
(video_stream_encoder.cc:1594): OnBitrateUpdated, bitrate 268000 stable bitrate = 268000 link allocation bitrate = 268000 packet loss 0 rtt 0
(video_stream_encoder.cc:1619): Video suspend state changed to: not suspended
(channel.cc:970): Changing video state, send=1
(video_stream_encoder.cc:1130): Encoder settings changed from EncoderInfo { ScalingSettings { min_pixels_per_frame = 57600 }, requested_resolution_alignment = 1, supports_native_handle = 0, implementation_name = 'unknown', has_trusted_rate_controller = 0, is_hardware_accelerated = 1, has_internal_source = 0, fps_allocation = [[ 1] ], resolution_bitrate_limits = [] , supports_simulcast = 0} to EncoderInfo { ScalingSettings { Thresholds { low = 29, high = 95}, min_pixels_per_frame = 57600 }, requested_resolution_alignment = 1, supports_native_handle = 0, implementation_name = 'libvpx', has_trusted_rate_controller = 0, is_hardware_accelerated = 0, has_internal_source = 0, fps_allocation = [[ 1] ], resolution_bitrate_limits = [] , supports_simulcast = 1}
// =========仅仅发了第一个视频包============
(rtp_sender_video.cc:667): Sent first RTP packet of the first video frame (pre-pacer)
(rtp_sender_video.cc:671): Sent last RTP packet of the first video frame (pre-pacer)
(video_send_stream_impl.cc:467): SignalEncoderActive, Encoder is activ//yin

采集音频线程的逻辑：（core_audio_base_win.cc）

void Run(void* obj) {  //启动线程 RTC_DCHECK(obj);reinterpret_cast<CoreAudioBase*>(obj)->ThreadRun();
}
void CoreAudioBase::ThreadRun() {   // 线程体
//...HANDLE wait_array[] = {stop_event_.Get(), restart_event_.Get(),audio_samples_event_.Get()};// Keep streaming audio until the stop event or the stream-switch event// is signaled. An error event can also break the main thread loop.while (streaming && !error) {   //死循环执行，直到有错，立即退出线程// Wait for a close-down event, stream-switch event or a new render event.DWORD wait_result = WaitForMultipleObjects(arraysize(wait_array),wait_array, false, INFINITE);switch (wait_result) {case WAIT_OBJECT_0 + 0:// |stop_event_| has been set.streaming = false;break;case WAIT_OBJECT_0 + 1:// |restart_event_| has been set.error = !HandleRestartEvent();break;case WAIT_OBJECT_0 + 2:{// |audio_samples_event_| has been set.error = !on_data_callback_(device_frequency);if(!initialized_ || !is_active_)   // 原来没有这里的判断条件，修改的这里{RTC_LOG(INFO) << "audio base not init, initialized:" << initialized_ << " is_active_:" << is_active_;error = 0;}break;}default:error = true;break;}}if (streaming && error) {  //退出的逻辑RTC_LOG(LS_ERROR) << "[" << DirectionToString(direction())<< "] WASAPI streaming failed. streaming:" << streaming<< " error:" << error;// Stop audio streaming since something has gone wrong in our main thread// loop. Note that, we are still in a "started" state, hence a Stop() call// is required to join the thread properly.result = audio_client_->Stop();if (FAILED(result.Error())) {RTC_LOG(LS_ERROR) << "IAudioClient::Stop failed: "<< core_audio_utility::ErrorToString(result);}// TODO(henrika): notify clients that something has gone wrong and that// this stream should be destroyed instead of reused in the future.}RTC_DLOG(INFO) << "[" << DirectionToString(direction())<< "] ...ThreadRun stops";
}
//回调数据的线程
bool CoreAudioInput::OnDataCallback(uint64_t device_frequency) {RTC_DCHECK_RUN_ON(&thread_checker_audio_);if (!initialized_ || !is_active_) {  // 还没有初始化，则直接返回false，这里使用全局变量来判断// This is concurrent examination of state across multiple threads so will// be somewhat error prone, but we should still be defensive and not use// audio_capture_client_ if we know it's not there.RTC_LOG(INFO) << "data call back, initialized:" << initialized_ << " is_active_:" << is_active_;return false;}   // ...
}

开启线程的逻辑：

CoreAudioBase::CoreAudioBase(Direction direction,bool automatic_restart,OnDataCallback data_callback,OnErrorCallback error_callback): format_(),direction_(direction),automatic_restart_(automatic_restart),on_data_callback_(data_callback),on_error_callback_(error_callback),device_index_(kUndefined),is_restarting_(false) {RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction) << "]";RTC_DLOG(INFO) << "Automatic restart: " << automatic_restart;RTC_DLOG(INFO) << "Windows version: " << rtc::rtc_win::GetVersion();// Create the event which the audio engine will signal each time a buffer// becomes ready to be processed by the client.// 从这里往下的三个条件变量要注意audio_samples_event_.Set(CreateEvent(nullptr, false, false, nullptr));RTC_DCHECK(audio_samples_event_.IsValid());// Event to be set in Stop() when rendering/capturing shall stop.stop_event_.Set(CreateEvent(nullptr, false, false, nullptr));RTC_DCHECK(stop_event_.IsValid());// Event to be set when it has been detected that an active device has been// invalidated or the stream format has changed.restart_event_.Set(CreateEvent(nullptr, false, false, nullptr));RTC_DCHECK(restart_event_.IsValid());enumerator_ = core_audio_utility::CreateDeviceEnumerator();enumerator_->RegisterEndpointNotificationCallback(this);RTC_LOG(INFO) << __FUNCTION__<< ":Registered endpoint notification callback.";
}

解决方案：

1.没有初始化成功不算返回报错，去掉这种错误。
2.如果上面的流程都是正确的，webrtc 的打印 log 显示音频通道正常，wireshark 发了音频 rtp 包，仍然听不到音频。可能是麦克风没有打卡，通话建立的时候往往会有一个麦克风的标志，检查下此标志是否打开。

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owt webrtc 音频没有声音

项目场景：

问题描述：

原因分析：

解决方案：

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