| 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;
|
| }
|
| }
|
|
|
|
|