Index: webrtc/voice_engine/channel_proxy.cc |
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc |
index 7f68f5101a180e45079cafb3fdceaa11f007ae5d..86501409e242ebc7d453e42fbd99b3da7163632d 100644 |
--- a/webrtc/voice_engine/channel_proxy.cc |
+++ b/webrtc/voice_engine/channel_proxy.cc |
@@ -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) { |