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Makes AudioSendStream signal that it's part of allocation.

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This adds calls to the underlying RtpRtcp module to indicate when audio
is part of bitrate allocation. This information is propagated and set in
the packet info for each packet.

This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.

Bug: webrtc:9796
Change-Id: I79b024cb7f2eb8c86421cfa34d38ef68467776c3
Reviewed-on: https://webrtc-review.googlesource.com/c/104882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25086}
This commit is contained in:
Sebastian Jansson
2018-10-10 10:23:13 +02:00
committed by Commit Bot
parent 99a70a2d78
commit b686396ec6
1 changed files with 5 additions and 0 deletions
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5
audio/audio_send_stream.cc
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@ -301,9 +301,12 @@ void AudioSendStream::Start() {
webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC"))) {
// Audio BWE is enabled.
transport_->packet_sender()->SetAccountForAudioPackets(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
config_.bitrate_priority,
has_transport_sequence_number);
} else {
rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
channel_proxy_->StartSend();
sending_ = true;
@ -713,8 +716,10 @@ void AudioSendStream::ReconfigureBitrateObserver(
stream->ConfigureBitrateObserver(
new_config.min_bitrate_bps, new_config.max_bitrate_bps,
new_config.bitrate_priority, has_transport_sequence_number);
stream->rtp_rtcp_module_->SetAsPartOfAllocation(true);
} else {
stream->RemoveBitrateObserver();
stream->rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
}