Support removing b=AS bandwidth constraints.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 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 697 matching lines @@
   // Handle RTP header extensions.
   std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
       params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
   if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) {
     changed_params->rtp_header_extensions =
         rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
   }
 
   // Handle max bitrate.
   if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps &&
-      params.max_bandwidth_bps >= 0) {
-    // 0 uncaps max bitrate (-1).
+      params.max_bandwidth_bps >= -1) {
+    // 0 or -1 uncaps max bitrate.
+    // TODO(pbos): Reconsider whether 0 should be invalid or not. -1 is the
+    // auto-bitrate value.

[Taylor Brandstetter, 2017/03/09 23:48:38]

The "correct" behavior for b=AS:0 may be "set bitrate to 0, don't send any
bits". I can't find any reference to 0 being a special value. But we should
change that separately.

[pbos-webrtc, 2017/03/10 18:52:09]

TODO text updated.

     changed_params->max_bandwidth_bps = rtc::Optional<int>(
         params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps);
   }
 
   // Handle conference mode.
   if (params.conference_mode != send_params_.conference_mode) {
     changed_params->conference_mode =
         rtc::Optional<bool>(params.conference_mode);
   }
 
@@ skipping 25 matching lines @@
     LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString();
   }
 
   if (changed_params.rtp_header_extensions) {
     send_rtp_extensions_ = changed_params.rtp_header_extensions;
   }
 
   if (changed_params.codec || changed_params.max_bandwidth_bps) {
     if (send_codec_) {
       // TODO(holmer): Changing the codec parameters shouldn't necessarily mean
-      // that we change the min/max of bandwidth estimation. Reevaluate this.
+      // that we change the min of bandwidth estimation. Reevaluate this.
       bitrate_config_ = GetBitrateConfigForCodec(send_codec_->codec);
       if (!changed_params.codec) {
         // 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;
       }
     }
-    if (params.max_bandwidth_bps >= 0) {
+    if (params.max_bandwidth_bps >= -1) {
       // Note that max_bandwidth_bps intentionally takes priority over the
       // bitrate config for the codec. This allows FEC to be applied above the
       // codec target bitrate.
       // TODO(pbos): Figure out whether b=AS means max bitrate for this
       // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC),
       // in which case this should not set a Call::BitrateConfig but rather
       // reconfigure all senders.
+
+      // TODO(pbos): Consider whether 0 should be invalid or not.
+      // 0 or -1 uncaps max bitrate.
       bitrate_config_.max_bitrate_bps =
           params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
     }
     call_->SetBitrateConfig(bitrate_config_);
   }
 
   {
     rtc::CritScope stream_lock(&stream_crit_);
     for (auto& kv : send_streams_) {
       kv.second->SetSendParameters(changed_params);
@@ skipping 1757 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             ulpfec_config.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket