Don't allocate audio if we have no transport sequence number.
Bug: chromium:1002875
Change-Id: I597184e59cf7b5f47b2025d26408069199ada2c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156305
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29432}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7f42d73..dbca457 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -360,12 +360,9 @@
if (sending_) {
return;
}
- // TODO(srte): We should not add audio to allocation just because
- // audio_send_side_bwe_ is false.
if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
config_.max_bitrate_bps != -1 &&
- (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0 ||
- !audio_send_side_bwe_)) {
+ (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
rtc::Event thread_sync_event;
@@ -824,11 +821,8 @@
return;
}
- // TODO(srte): We should not add audio to allocation just because
- // audio_send_side_bwe_ is false.
if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
- new_config.max_bitrate_bps != -1 &&
- (TransportSeqNumId(new_config) != 0 || !audio_send_side_bwe_)) {
+ new_config.max_bitrate_bps != -1 && TransportSeqNumId(new_config) != 0) {
rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] {
