Move some send stream configuration into webrtc::AudioSendStream.

webrtc/audio/audio_send_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.
  */
 
@@ skipping 16 matching lines @@
   std::stringstream ss;
   ss << "{ssrc: " << ssrc;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
     if (i != extensions.size() - 1) {
       ss << ", ";
     }
   }
   ss << ']';
+  ss << ", c_name: " << c_name;
   ss << '}';
   return ss.str();
 }
 
 std::string AudioSendStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", voe_channel_id: " << voe_channel_id;
   // TODO(solenberg): Encoder config.
   ss << ", cng_payload_type: " << cng_payload_type;
   ss << ", red_payload_type: " << red_payload_type;
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
     : config_(config), audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
+
+  const int channel_id = config.voe_channel_id;
+  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
+  int error = rtp->SetRTCPStatus(channel_id, true);
+  RTC_DCHECK_EQ(0, error);
+  error = rtp->SetLocalSSRC(channel_id, config.rtp.ssrc);
+  RTC_DCHECK_EQ(0, error);
+  error = rtp->SetRTCP_CNAME(channel_id, config.rtp.c_name.c_str());
+  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::kAbsSendTime) {
+      error = rtp->SetSendAbsoluteSenderTimeStatus(channel_id, true,
+                                                   extension.id);
+      RTC_DCHECK_EQ(0, error);
+    } else if (extension.name == RtpExtension::kAudioLevel) {
+      error = rtp->SetSendAudioLevelIndicationStatus(channel_id, true,
+                                                     extension.id);
+      RTC_DCHECK_EQ(0, error);
+    } else {
+      RTC_NOTREACHED() << "Registering unsupported RTP extension.";
+    }
+  }
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
 }
 
+void AudioSendStream::Start() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+void AudioSendStream::Stop() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+void AudioSendStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  return false;
+}
+
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
-  internal::AudioState* audio_state =
-      static_cast<internal::AudioState*>(audio_state_.get());
-  VoiceEngine* voice_engine = audio_state->voice_engine();
-  ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine);
-  ScopedVoEInterface<VoECodec> codec(voice_engine);
-  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine);
-  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine);
+  ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
+  ScopedVoEInterface<VoECodec> codec(voice_engine());
+  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
+  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
   unsigned int ssrc = 0;
   webrtc::CallStatistics call_stats = {0};
+  // TODO(solenberg): Change error code checking to RTC_CHECK_EQ(..., -1), if
+  //                  possible...
   if (rtp->GetLocalSSRC(config_.voe_channel_id, ssrc) == -1 ||
       rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1) {
     return stats;
   }
 
   stats.bytes_sent = call_stats.bytesSent;
   stats.packets_sent = call_stats.packetsSent;
 
   webrtc::CodecInst codec_inst = {0};
   if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
@@ skipping 55 matching lines @@
     int erl = -100;
     int erle = -100;
     int dummy1 = 0;
     int dummy2 = 0;
     if (processing->GetEchoMetrics(erl, erle, dummy1, dummy2) != -1) {
       stats.echo_return_loss = erl;
       stats.echo_return_loss_enhancement = erle;
     }
   }
 
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
   stats.typing_noise_detected = audio_state->typing_noise_detected();
 
   return stats;
 }
 
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
-void AudioSendStream::Start() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-}
-
-void AudioSendStream::Stop() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-}
-
-void AudioSendStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-}
-
-bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  // TODO(solenberg): Tests call this function on a network thread, libjingle
-  // calls on the worker thread. We should move towards always using a network
-  // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  return false;
+VoiceEngine* AudioSendStream::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