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}
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@ -360,12 +360,9 @@ void AudioSendStream::Start() {
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if (sending_) {
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if (sending_) {
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return;
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return;
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}
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}
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// TODO(srte): We should not add audio to allocation just because
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// audio_send_side_bwe_ is false.
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if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
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if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
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config_.max_bitrate_bps != -1 &&
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config_.max_bitrate_bps != -1 &&
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(allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0 ||
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(allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
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!audio_send_side_bwe_)) {
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rtp_transport_->AccountForAudioPacketsInPacedSender(true);
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rtp_transport_->AccountForAudioPacketsInPacedSender(true);
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rtp_rtcp_module_->SetAsPartOfAllocation(true);
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rtp_rtcp_module_->SetAsPartOfAllocation(true);
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rtc::Event thread_sync_event;
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rtc::Event thread_sync_event;
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@ -824,11 +821,8 @@ void AudioSendStream::ReconfigureBitrateObserver(
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return;
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return;
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}
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}
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// TODO(srte): We should not add audio to allocation just because
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// audio_send_side_bwe_ is false.
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if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
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if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
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new_config.max_bitrate_bps != -1 &&
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new_config.max_bitrate_bps != -1 && TransportSeqNumId(new_config) != 0) {
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(TransportSeqNumId(new_config) != 0 || !audio_send_side_bwe_)) {
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rtp_transport_->AccountForAudioPacketsInPacedSender(true);
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rtp_transport_->AccountForAudioPacketsInPacedSender(true);
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rtc::Event thread_sync_event;
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rtc::Event thread_sync_event;
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worker_queue_->PostTask([&] {
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worker_queue_->PostTask([&] {
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