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Unified Diff: talk/media/webrtc/webrtcvoiceengine.cc

Issue 1347353004: Reduce LS_INFO spam from voice_engine/. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 642fd3d233e649ecbcb989a85391176b1761e0dc..0feb0fb6fa1d547f437cb88d76ff599c464993b0 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1242,34 +1242,6 @@ void WebRtcVoiceEngine::SetTraceOptions(const std::string& options) {
}
}
-// Ignore spammy trace messages, mostly from the stats API when we haven't
-// gotten RTCP info yet from the remote side.
-bool WebRtcVoiceEngine::ShouldIgnoreTrace(const std::string& trace) {
- static const char* kTracesToIgnore[] = {
- "\tfailed to GetReportBlockInformation",
- "GetRecCodec() failed to get received codec",
- "GetReceivedRtcpStatistics: Could not get received RTP statistics",
- "GetRemoteRTCPData() failed to measure statistics due to lack of received RTP and/or RTCP packets", // NOLINT
- "GetRemoteRTCPData() failed to retrieve sender info for remote side",
- "GetRTPStatistics() failed to measure RTT since no RTP packets have been received yet", // NOLINT
- "GetRTPStatistics() failed to read RTP statistics from the RTP/RTCP module",
- "GetRTPStatistics() failed to retrieve RTT from the RTP/RTCP module",
- "SenderInfoReceived No received SR",
- "StatisticsRTP() no statistics available",
- "TransmitMixer::TypingDetection() VE_TYPING_NOISE_WARNING message has been posted", // NOLINT
- "TransmitMixer::TypingDetection() pending noise-saturation warning exists", // NOLINT
- "GetRecPayloadType() failed to retrieve RX payload type (error=10026)", // NOLINT
- "StopPlayingFileAsMicrophone() isnot playing (error=8088)",
- NULL
- };
- for (const char* const* p = kTracesToIgnore; *p; ++p) {
- if (trace.find(*p) != std::string::npos) {
- return true;
- }
- }
- return false;
-}
-
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
int length) {
rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
@@ -1289,9 +1261,7 @@ void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
LOG_V(sev) << msg;
} else {
std::string msg(trace + 71, length - 72);
- if (!ShouldIgnoreTrace(msg)) {
- LOG_V(sev) << "webrtc: " << msg;
- }
+ LOG_V(sev) << "webrtc: " << msg;
}
}
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