Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
     if (i != extensions.size() - 1) {
@@ skipping 30 matching lines @@
     : remote_bitrate_estimator_(remote_bitrate_estimator),
       config_(config),
       audio_state_(audio_state),
       rtp_header_parser_(RtpHeaderParser::Create()) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(remote_bitrate_estimator_);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(rtp_header_parser_);
 
+  VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
+  channel_proxy_.reset(voe_impl->GetChannelProxy(config_.voe_channel_id));
+  channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
+
   const int channel_id = config.voe_channel_id;
   ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
-  int error = rtp->SetLocalSSRC(channel_id, config.rtp.local_ssrc);
-  RTC_DCHECK_EQ(0, error);
   for (const auto& extension : config.rtp.extensions) {
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(extension.id, 1);
     RTC_DCHECK_LE(extension.id, 14);
     if (extension.name == RtpExtension::kAudioLevel) {
-      error = rtp->SetReceiveAudioLevelIndicationStatus(channel_id, true,
-                                                        extension.id);
+      int error = rtp->SetReceiveAudioLevelIndicationStatus(channel_id, true,
+                                                            extension.id);
       RTC_DCHECK_EQ(0, error);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, extension.id);
       RTC_DCHECK(registered);
     } else if (extension.name == RtpExtension::kAbsSendTime) {
-      error = rtp->SetReceiveAbsoluteSenderTimeStatus(channel_id, true,
-                                                      extension.id);
+      int error = rtp->SetReceiveAbsoluteSenderTimeStatus(channel_id, true,
+                                                          extension.id);
       RTC_DCHECK_EQ(0, error);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, extension.id);
       RTC_DCHECK(registered);
     } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
       // TODO(holmer): Need to do something here or in  DeliverRtp() to actually
       //               handle audio packets with this header extension.
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
@@ skipping 131 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc