Build WebRTC with data channel only.

webrtc/pc/channel.cc

 /*
  *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <utility>
 
 #include "webrtc/pc/channel.h"
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/rtputils.h"
-#include "webrtc/media/engine/webrtcvoiceengine.h"
+// Adding 'nogncheck' to disable the gn include headers check to support modular
+// WebRTC build targets.
+#include "webrtc/media/engine/webrtcvoiceengine.h"  // nogncheck
 #include "webrtc/p2p/base/packettransportinternal.h"
 #include "webrtc/pc/channelmanager.h"
 
 namespace cricket {
 using rtc::Bind;
 
 namespace {
 // See comment below for why we need to use a pointer to a unique_ptr.
 bool SetRawAudioSink_w(VoiceMediaChannel* channel,
                        uint32_t ssrc,
@@ skipping 1526 matching lines @@
   return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
   return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
                                             media_channel(), stats));
 }
 
 std::vector<webrtc::RtpSource> VoiceChannel::GetSources(uint32_t ssrc) const {
   return worker_thread()->Invoke<std::vector<webrtc::RtpSource>>(
-      RTC_FROM_HERE,
-      Bind(&WebRtcVoiceMediaChannel::GetSources,
-           static_cast<WebRtcVoiceMediaChannel*>(media_channel()), ssrc));
+      RTC_FROM_HERE, Bind(&VoiceChannel::GetSources_w, this, ssrc));
+}
+
+std::vector<webrtc::RtpSource> VoiceChannel::GetSources_w(uint32_t ssrc) const {
+  RTC_DCHECK(worker_thread()->IsCurrent());
+  return media_channel()->GetSources(ssrc);
 }
 
 void VoiceChannel::StartMediaMonitor(int cms) {
   media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
       rtc::Thread::Current()));
   media_monitor_->SignalUpdate.connect(
       this, &VoiceChannel::OnMediaMonitorUpdate);
   media_monitor_->Start(cms);
 }
 
@@ skipping 774 matching lines @@
 
 void RtpDataChannel::OnDataChannelReadyToSend(bool writable) {
   // This is usded for congestion control to indicate that the stream is ready
   // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
   // that the transport channel is ready.
   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
                            new DataChannelReadyToSendMessageData(writable));
 }
 
 }  // namespace cricket