Support for remote audio into tracks

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 26 matching lines @@
 #include <cstdio>
 #include <string>
 #include <vector>
 
 #include "talk/media/base/audioframe.h"
 #include "talk/media/base/audiorenderer.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/streamparams.h"
 #include "talk/media/webrtc/webrtcmediaengine.h"
 #include "talk/media/webrtc/webrtcvoe.h"
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
@@ skipping 1184 matching lines @@
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     return stream_->GetStats();
   }
 
   int channel() const {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     return config_.voe_channel_id;
   }
 
+  void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+    stream_->SetSink(std::move(sink));
+  }
+
  private:
   void RecreateAudioReceiveStream(bool use_combined_bwe,
       const std::vector<webrtc::RtpExtension>& extensions) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioReceiveStream(stream_);
       stream_ = nullptr;
     }
     config_.rtp.extensions = extensions;
     config_.combined_audio_video_bwe = use_combined_bwe;
@@ skipping 764 matching lines @@
   // Deregister default channel, if that's the one being destroyed.
   if (IsDefaultRecvStream(ssrc)) {
     default_recv_ssrc_ = -1;
   }
 
   const int channel = it->second->channel();
 
   // Clean up and delete the receive stream+channel.
   LOG(LS_INFO) << "Removing audio receive stream " << ssrc
                << " with VoiceEngine channel #" << channel << ".";
+  it->second->SetRawAudioSink(nullptr);
   delete it->second;
   recv_streams_.erase(it);
   return DeleteVoEChannel(channel);
 }
 
 bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc,
                                                AudioRenderer* renderer) {
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
     if (renderer) {
@@ skipping 356 matching lines @@
     rinfo.decoding_plc = stats.decoding_plc;
     rinfo.decoding_cng = stats.decoding_cng;
     rinfo.decoding_plc_cng = stats.decoding_plc_cng;
     rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
     info->receivers.push_back(rinfo);
   }
 
   return true;
 }
 
+void WebRtcVoiceMediaChannel::SetRawAudioSink(
+    uint32_t ssrc,
+    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink";
+  const auto it = recv_streams_.find(ssrc);
+  if (it == recv_streams_.end()) {
+    LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
+    return;
+  }
+  it->second->SetRawAudioSink(std::move(sink));
+}
+
 int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
   unsigned int ulevel = 0;
   int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
   return (ret == 0) ? static_cast<int>(ulevel) : -1;
 }
 
 int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = recv_streams_.find(ssrc);
   if (it != recv_streams_.end()) {
@@ skipping 64 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE