Added a flag to AudioCodecSpec to indicate adaptive bitrate support.

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 1148 matching lines @@
   // Only generate CN payload types for these clockrates:
   std::map<int, bool, std::greater<int>> generate_cn = {{ 8000,  false },
                                                         { 16000, false },
                                                         { 32000, false }};
   // Only generate telephone-event payload types for these clockrates:
   std::map<int, bool, std::greater<int>> generate_dtmf = {{ 8000,  false },
                                                           { 16000, false },
                                                           { 32000, false },
                                                           { 48000, false }};
 
-  auto map_format = [&mapper, &out] (const webrtc::SdpAudioFormat& format) {
+  auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
+                              AudioCodecs* out) {
     rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
-    if (!opt_codec) {
+    if (opt_codec) {
+      if (out) {
+        out->push_back(*opt_codec);
+      }
+    } else {
       LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
-      return false;
     }
 
-    auto& codec = *opt_codec;
-    if (IsCodec(codec, kOpusCodecName)) {
-      // TODO(ossu): Set this specifically for Opus for now, until we have a
-      // better way of dealing with rtcp-fb parameters.
-      codec.AddFeedbackParam(
-          FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
-    }
-    out.push_back(codec);
-    return true;
+    return opt_codec;
   };
 
   for (const auto& spec : specs) {
-    if (map_format(spec.format)) {
+    // We need to do some extra stuff before adding the main codecs to out.
+    rtc::Optional<AudioCodec> opt_codec = map_format(spec.format, nullptr);
+    if (opt_codec) {
+      AudioCodec& codec = *opt_codec;
+      if (spec.supports_network_adaption) {
+        codec.AddFeedbackParam(
+            FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
+      }
+
       if (spec.allow_comfort_noise) {
         // Generate a CN entry if the decoder allows it and we support the
         // clockrate.
         auto cn = generate_cn.find(spec.format.clockrate_hz);
         if (cn != generate_cn.end()) {
           cn->second = true;
         }
       }
 
       // Generate a telephone-event entry if we support the clockrate.
       auto dtmf = generate_dtmf.find(spec.format.clockrate_hz);
       if (dtmf != generate_dtmf.end()) {
         dtmf->second = true;
       }
+
+      out.push_back(codec);
     }
   }
 
   // Add CN codecs after "proper" audio codecs.
   for (const auto& cn : generate_cn) {
     if (cn.second) {
-      map_format({kCnCodecName, cn.first, 1});
+      map_format({kCnCodecName, cn.first, 1}, &out);
     }
   }
 
   // Add telephone-event codecs last.
   for (const auto& dtmf : generate_dtmf) {
     if (dtmf.second) {
-      map_format({kDtmfCodecName, dtmf.first, 1});
+      map_format({kDtmfCodecName, dtmf.first, 1}, &out);
     }
   }
 
   return out;
 }
 
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioSource::Sink {
  public:
   WebRtcAudioSendStream(
@@ skipping 1492 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