Default enable sending transport sequence numbers on audio packets.
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.
Transport-cc extension still needs to be negotiated.
Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 1c0a32f..4e21b1f 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -143,7 +143,6 @@
std::unique_ptr<voe::ChannelSendInterface> channel_send)
: clock_(clock),
worker_queue_(rtp_transport->GetWorkerQueue()),
- audio_send_side_bwe_(field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")),
allocate_audio_without_feedback_(
field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")),
enable_audio_alr_probing_(
@@ -289,7 +288,7 @@
RtcpBandwidthObserver* bandwidth_observer = nullptr;
- if (audio_send_side_bwe_ && !allocate_audio_without_feedback_ &&
+ if (!allocate_audio_without_feedback_ &&
new_ids.transport_sequence_number != 0) {
rtp_rtcp_module_->RegisterRtpHeaderExtension(
TransportSequenceNumber::kUri, new_ids.transport_sequence_number);
@@ -809,8 +808,7 @@
if (config_.min_bitrate_bps == new_config.min_bitrate_bps &&
config_.max_bitrate_bps == new_config.max_bitrate_bps &&
config_.bitrate_priority == new_config.bitrate_priority &&
- (TransportSeqNumId(config_) == TransportSeqNumId(new_config) ||
- !audio_send_side_bwe_) &&
+ TransportSeqNumId(config_) == TransportSeqNumId(new_config) &&
config_.audio_network_adaptor_config ==
new_config.audio_network_adaptor_config) {
return;