Log audio stream start/stop.
Bug: None Change-Id: I3f97672bc7d29dd6023fe8b6bdf98d699622841d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331160 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41362}
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@ -183,6 +183,7 @@ void AudioReceiveStreamImpl::Start() {
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if (playing_) {
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return;
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}
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RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Start: " << remote_ssrc();
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channel_receive_->StartPlayout();
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playing_ = true;
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audio_state()->AddReceivingStream(this);
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@ -193,6 +194,7 @@ void AudioReceiveStreamImpl::Stop() {
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if (!playing_) {
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return;
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}
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RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Stop: " << remote_ssrc();
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channel_receive_->StopPlayout();
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playing_ = false;
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audio_state()->RemoveReceivingStream(this);
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@ -355,6 +355,7 @@ void AudioSendStream::Start() {
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if (sending_) {
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return;
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}
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RTC_LOG(LS_INFO) << "AudioSendStream::Start: " << config_.rtp.ssrc;
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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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(allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
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@ -376,7 +377,7 @@ void AudioSendStream::Stop() {
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if (!sending_) {
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return;
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}
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RTC_LOG(LS_INFO) << "AudioSendStream::Stop: " << config_.rtp.ssrc;
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RemoveBitrateObserver();
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channel_send_->StopSend();
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sending_ = false;
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