commit bbbe4e1a15edbbfe99bb13f69593414865f720f2
author Alex Narest <alexnarest@webrtc.org> Fri Jul 13 08:32:58 2018
committer Commit Bot <commit-bot@chromium.org> Fri Jul 13 09:06:06 2018
tree 00296e593d15fd96e90054631ca0f05fd521e259
parent e9a18b263570b8478207e2006a448c61b3f6655b

Better handle target audio bitrate allocation.

Do not update target audio bitrate if WebRTC-Audio-SendSideBwe-For-Video is enabled but other side does not support TWCC

Bug: webrtc:8243
Change-Id: I6c3c4f223dc5168d726996324717d7ba9ec96e6c
Reviewed-on: https://webrtc-review.googlesource.com/88440
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23963}

audio/audio_send_stream.cc

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 9fdf5f6..de1be2c 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -525,9 +525,17 @@
                        << rtc::ToString(spec.format);
     return false;
   }
+
+  // If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is
+  // not enabled, do not update target audio bitrate if we are in
+  // WebRTC-Audio-SendSideBwe-For-Video experiment
+  const bool do_not_update_target_bitrate =
+      !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") &&
+      webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") &&
+      !FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
   // If a bitrate has been specified for the codec, use it over the
   // codec's default.
-  if (spec.target_bitrate_bps) {
+  if (!do_not_update_target_bitrate && spec.target_bitrate_bps) {
     encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps);
   }
 
@@ -590,11 +598,19 @@
     return SetupSendCodec(stream, new_config);
   }
 
+  // If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is
+  // not enabled, do not update target audio bitrate if we are in
+  // WebRTC-Audio-SendSideBwe-For-Video experiment
+  const bool do_not_update_target_bitrate =
+      !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") &&
+      webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") &&
+      !FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
+
   const absl::optional<int>& new_target_bitrate_bps =
       new_config.send_codec_spec->target_bitrate_bps;
   // If a bitrate has been specified for the codec, use it over the
   // codec's default.
-  if (new_target_bitrate_bps &&
+  if (!do_not_update_target_bitrate && new_target_bitrate_bps &&
       new_target_bitrate_bps !=
           old_config.send_codec_spec->target_bitrate_bps) {
     CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {