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}

audio/audio_send_stream.cc

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;