On WVoMC::SetSendParameters(), figure out send codec settings ONCE, not for each send stream.

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 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.
  */
 
@@ skipping 214 matching lines @@
   void SetNack(int channel, bool nack_enabled);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool ChangePlayout(bool playout);
   bool ChangeSend(SendFlags send);
   bool ChangeSend(int channel, SendFlags send);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
-  bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
+  bool SetSendCodecs(int channel);
   bool SetSendBitrateInternal(int bps);
+  bool HasSendCodec() const {
+    return send_codec_spec_.codec_inst.pltype != -1;
+  }
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
-  std::vector<AudioCodec> send_codecs_;
-  std::unique_ptr<webrtc::CodecInst> send_codec_;
   bool send_bitrate_setting_ = false;
   int send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;
-  bool nack_enabled_ = false;
-  bool transport_cc_enabled_ = false;
+  bool recv_transport_cc_enabled_ = false;
   bool playout_ = false;
   SendFlags desired_send_ = SEND_NOTHING;
   SendFlags send_ = SEND_NOTHING;
   webrtc::Call* const call_ = nullptr;
 
   // SSRC of unsignalled receive stream, or -1 if there isn't one.
   int64_t default_recv_ssrc_ = -1;
   // Volume for unsignalled stream, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
   // Sink for unsignalled stream, which may be set before the stream exists.
   std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
   // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
   // and https://code.google.com/p/chromium/issues/detail?id=547661
   uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
+  struct SendCodecSpec {
+    SendCodecSpec() {
+      std::memset(&codec_inst, 0, sizeof(codec_inst));

[ossu, 2016/03/04 14:40:22]

I believe CodecInst could be zero initialized with = {0} here as well (see
comments in the .cc file). Some say = {} should work as well. Look it up! :) I
know I will.

[the sun, 2016/03/04 15:44:03]

Yes, unfortunately direct braced init of the field won't work though.

+      codec_inst.pltype = -1;
+    }
+    bool nack_enabled = false;
+    bool transport_cc_enabled = false;
+    bool enable_codec_fec = false;
+    bool enable_opus_dtx = false;
+    int opus_max_playback_rate = 0;
+    int red_payload_type = -1;
+    int cng_payload_type = -1;
+    int cng_plfreq = -1;
+    webrtc::CodecInst codec_inst;
+  } send_codec_spec_;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_