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Fix flaky test WebRtcMediaRecorderTest.PeerConnection

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A previous CL changed from logging to DCHECKing when setting minimum playout delay on a VoE channel: https://codereview.webrtc.org/2452163004/

I thought it safe at the time, since the input parameter range is capped, but apparently I didn't dig deep enough, as ultimately a failure may be returned for other reasons: https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/audio_coding/neteq/delay_manager.cc#381

Thus, reverting to old behavior.

BUG=694373

Review-Url: https://codereview.webrtc.org/2704933008
Cr-Commit-Position: refs/heads/master@{#16775}
This commit is contained in:
solenberg 2017-02-22 07:07:04 -08:00 committed by Commit bot
parent 1d4e3d8a2e
commit 0335e6c4bf
1 changed files with 4 additions and 1 deletions
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5
webrtc/voice_engine/channel_proxy.cc
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@ -14,6 +14,7 @@
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/voice_engine/channel.h"
namespace webrtc {
@ -284,7 +285,9 @@ void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
// close as possible, instead of failing.
delay_ms = std::max(0, std::min(delay_ms, 10000));
int error = channel()->SetMinimumPlayoutDelay(delay_ms);
RTC_DCHECK_EQ(0, error);
if (0 != error) {
LOG(LS_WARNING) << "Error setting minimum playout delay.";
}
}
void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {