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Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily.

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BUG=b/34746131

Review-Url: https://codereview.webrtc.org/2670183005
Cr-Commit-Position: refs/heads/master@{#16456}
This commit is contained in:
solenberg 2017-02-06 13:03:19 -08:00 committed by Commit bot
parent bd9a77f4e5
commit f748ca47b0
1 changed files with 7 additions and 8 deletions
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15
webrtc/media/engine/webrtcvoiceengine.cc
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@ -2499,14 +2499,6 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
return;
}
if (default_recv_ssrc_ != -1) {
LOG(LS_INFO) << "Removing default receive stream with ssrc "
<< default_recv_ssrc_;
RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
RemoveRecvStream(default_recv_ssrc_);
default_recv_ssrc_ = -1;
}
StreamParams sp;
sp.ssrcs.push_back(ssrc);
LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
@ -2514,7 +2506,14 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
LOG(LS_WARNING) << "Could not create default receive stream.";
return;
}
if (default_recv_ssrc_ != -1) {
LOG(LS_INFO) << "Removing default receive stream with ssrc "
<< default_recv_ssrc_;
RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
RemoveRecvStream(default_recv_ssrc_);
}
default_recv_ssrc_ = ssrc;
SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
if (default_sink_) {
std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(