Injectable audio encoders: voice_engine/channel changes.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 1184 matching lines @@
   if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
         "StartSend() RTP/RTCP failed to stop sending");
   }
   _rtpRtcpModule->SetSendingMediaStatus(false);
 
   return 0;
 }
 
+bool Channel::SetSendFormat(int payload_type,
+                            const SdpAudioFormat& format,
+                            AudioEncoderFactory* factory) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::SetSendFormat()");
+
+  auto encoder = factory->MakeAudioEncoder(payload_type, format);
+  if (!encoder) {
+    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
+                 "SetSendFormat() failed to create encoder");
+    return false;
+  }
+
+  // TODO(ossu): Make a CodecInst up for now.

[the sun, 2017/02/22 14:08:38]

You need to explain why we should accept your lies. This is not politics.

[ossu, 2017/02/22 14:24:41]

Alright. I'll elaborate and give it a less offensive name. :)

+  CodecInst lies;
+  lies.pltype = payload_type;
+  strncpy(lies.plname, format.name.c_str(), sizeof(lies.plname));
+  lies.plname[sizeof(lies.plname) - 1] = 0;
+  // Seems unclear if it should be clock rate or sample rate. CodecInst
+  // supposedly carries the sample rate, but only clock rate seems sensible to
+  // send to the RTP/RTCP module.
+  lies.plfreq = format.clockrate_hz;
+  //lies.plfreq = encoder->SampleRateHz();
+  lies.pacsize = 0;
+  lies.channels = encoder->NumChannels();
+  lies.rate = 0;
+
+  if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
+    _rtpRtcpModule->DeRegisterSendPayload(payload_type);
+    if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) {
+      WEBRTC_TRACE(
+          kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
+          "SetSendFormat() failed to register codec to RTP/RTCP module");
+      return false;
+    }
+  }
+
+  audio_coding_->SetEncoder(std::move(encoder));
+
+  return true;
+}
+
 int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterVoiceEngineObserver()");
   rtc::CritScope cs(&_callbackCritSect);
 
   if (_voiceEngineObserverPtr) {
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceError,
         "RegisterVoiceEngineObserver() observer already enabled");
     return -1;
@@ skipping 1796 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc