Fix issue with conflicting behavior if setting a max BW with b=AS on both audio and video.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  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 1649 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
                << params.ToString();
   // TODO(pthatcher): Refactor this to be more clean now that we have
   // all the information at once.
 
   if (!SetSendCodecs(params.codecs)) {
     return false;
   }
 
-  if (params.max_bandwidth_bps >= 0) {
-    // Note that max_bandwidth_bps intentionally takes priority over the
-    // bitrate config for the codec.
-    bitrate_config_.max_bitrate_bps =
-        params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
-  }
-  call_->SetBitrateConfig(bitrate_config_);
-
   if (!ValidateRtpExtensions(params.extensions)) {
     return false;
   }
   std::vector<webrtc::RtpExtension> filtered_extensions =
       FilterRtpExtensions(params.extensions,
                           webrtc::RtpExtension::IsSupportedForAudio, true);
   if (send_rtp_extensions_ != filtered_extensions) {
     send_rtp_extensions_.swap(filtered_extensions);
     for (auto& it : send_streams_) {
       it.second->RecreateAudioSendStream(send_rtp_extensions_);
@@ skipping 251 matching lines @@
         dtmf_payload_freq_ = codec.clockrate;
       }
     }
   }
 
   // Scan through the list to figure out the codec to use for sending, along
   // with the proper configuration for VAD, CNG, NACK and Opus-specific
   // parameters.
   // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
+  webrtc::Call::Config::BitrateConfig bitrate_config;
   {
     send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
 
     // Find send codec (the first non-telephone-event/CN codec).
     const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
         codecs, &send_codec_spec.codec_inst);
     if (!codec) {
       LOG(LS_WARNING) << "Received empty list of codecs.";
       return false;
     }
 
     send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
     send_codec_spec.nack_enabled = HasNack(*codec);
-    bitrate_config_ = GetBitrateConfigForCodec(*codec);
+    bitrate_config = GetBitrateConfigForCodec(*codec);
 
     // For Opus as the send codec, we are to determine inband FEC, maximum
     // playback rate, and opus internal dtx.
     if (IsCodec(*codec, kOpusCodecName)) {
       GetOpusConfig(*codec, &send_codec_spec.codec_inst,
                     &send_codec_spec.enable_codec_fec,
                     &send_codec_spec.opus_max_playback_rate,
                     &send_codec_spec.enable_opus_dtx,
                     &send_codec_spec.min_ptime_ms,
                     &send_codec_spec.max_ptime_ms);
@@ skipping 55 matching lines @@
 
   if (send_codec_spec_ != send_codec_spec) {
     send_codec_spec_ = std::move(send_codec_spec);
     // Apply new settings to all streams.
     for (const auto& kv : send_streams_) {
       kv.second->RecreateAudioSendStream(send_codec_spec_);
     }
   } else {
     // If the codec isn't changing, set the start bitrate to -1 which means
     // "unchanged" so that BWE isn't affected.
-    bitrate_config_.start_bitrate_bps = -1;
+    bitrate_config.start_bitrate_bps = -1;
   }
+  call_->SetBitrateConfig(bitrate_config);
 
   // Check if the transport cc feedback or NACK status has changed on the
   // preferred send codec, and in that case reconfigure all receive streams.
   if (recv_transport_cc_enabled_ != send_codec_spec_.transport_cc_enabled ||
       recv_nack_enabled_ != send_codec_spec_.nack_enabled) {
     LOG(LS_INFO) << "Recreate all the receive streams because the send "
                     "codec has changed.";
     recv_transport_cc_enabled_ = send_codec_spec_.transport_cc_enabled;
     recv_nack_enabled_ = send_codec_spec_.nack_enabled;
     for (auto& kv : recv_streams_) {
@@ skipping 610 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE