From cd0eedb2483b8a1cb07c953f0c06aeec8ce6f144 Mon Sep 17 00:00:00 2001 From: Sebastian Jansson Date: Thu, 10 Oct 2019 13:52:26 +0200 Subject: [PATCH] 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 Reviewed-by: Oskar Sundbom Commit-Queue: Sebastian Jansson Cr-Commit-Position: refs/heads/master@{#29432} --- audio/audio_send_stream.cc | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index 7f42d73ad3..dbca457636 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -360,12 +360,9 @@ void AudioSendStream::Start() { 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 @@ void AudioSendStream::ReconfigureBitrateObserver( 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([&] {