WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/modules/audio_coding/neteq/neteq_impl.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 194 matching lines @@
   last_vad_activity_ = audio_frame->vad_activity_;
   last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
   RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
              last_output_sample_rate_hz_ == 16000 ||
              last_output_sample_rate_hz_ == 32000 ||
              last_output_sample_rate_hz_ == 48000)
       << "Unexpected sample rate " << last_output_sample_rate_hz_;
   return kOK;
 }
 
+void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  rtc::CritScope lock(&crit_sect_);
+  const std::vector<int> changed_payload_types =
+      decoder_database_->SetCodecs(codecs);
+  for (const int pt : changed_payload_types) {
+    packet_buffer_->DiscardPacketsWithPayloadType(pt);
+  }
+}
+
 int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
                                    const std::string& name,
                                    uint8_t rtp_payload_type) {
   rtc::CritScope lock(&crit_sect_);
   LOG(LS_VERBOSE) << "RegisterPayloadType "
                   << static_cast<int>(rtp_payload_type) << " "
                   << static_cast<int>(codec);
   int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec, name);
   if (ret != DecoderDatabase::kOK) {
     switch (ret) {
@@ skipping 1915 matching lines @@
   }
 }
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(
       fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
       *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
       tick_timer_.get()));
 }
 }  // namespace webrtc