From fcd12b3b7d7e92d6f8cdfdf8277808ae52a07c36 Mon Sep 17 00:00:00 2001 From: "wu@webrtc.org" Date: Thu, 15 Sep 2011 20:49:50 +0000 Subject: [PATCH] Add necessary spaces to log. Review URL: http://webrtc-codereview.appspot.com/148002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@602 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/voice_engine/main/source/channel.cc | 36 ++++++++++++------------- 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc index c89cdaab4..3d3032a50 100644 --- a/src/voice_engine/main/source/channel.cc +++ b/src/voice_engine/main/source/channel.cc @@ -506,7 +506,7 @@ Channel::OnReceivedTelephoneEvent(const WebRtc_Word32 id, { WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::OnReceivedTelephoneEvent(id=%d, event=%u," - "endOfEvent=%d)", id, event, endOfEvent); + " endOfEvent=%d)", id, event, endOfEvent); #ifdef WEBRTC_DTMF_DETECTION if (_outOfBandTelephoneEventDetecion) @@ -530,7 +530,7 @@ Channel::OnPlayTelephoneEvent(const WebRtc_Word32 id, { WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u," - "volume=%u)", id, event, lengthMs, volume); + " volume=%u)", id, event, lengthMs, volume); if (!_playOutbandDtmfEvent || (event > 15)) { @@ -1588,7 +1588,7 @@ Channel::Init() _engineStatisticsPtr->SetLastError( VE_APM_ERROR, kTraceWarning, "Channel::Init() failed to set the device sample rate to 48K" - "for far-end AP module"); + " for far-end AP module"); } if (_rxAudioProcessingModulePtr->set_sample_rate_hz(8000)) @@ -1612,7 +1612,7 @@ Channel::Init() _engineStatisticsPtr->SetLastError( VE_SOUNDCARD_ERROR, kTraceWarning, "Init() failed to set channels for the primary audio" - "stream"); + " stream"); } if (_rxAudioProcessingModulePtr->high_pass_filter()->Enable( @@ -1621,7 +1621,7 @@ Channel::Init() _engineStatisticsPtr->SetLastError( VE_APM_ERROR, kTraceWarning, "Channel::Init() failed to set the high-pass filter for" - "far-end AP module"); + " far-end AP module"); } if (_rxAudioProcessingModulePtr->noise_suppression()->set_level( @@ -1630,7 +1630,7 @@ Channel::Init() _engineStatisticsPtr->SetLastError( VE_APM_ERROR, kTraceWarning, "Init() failed to set noise reduction level for far-end" - "AP module"); + " AP module"); } if (_rxAudioProcessingModulePtr->noise_suppression()->Enable( WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE) != 0) @@ -1638,7 +1638,7 @@ Channel::Init() _engineStatisticsPtr->SetLastError( VE_APM_ERROR, kTraceWarning, "Init() failed to set noise reduction state for far-end" - "AP module"); + " AP module"); } if (_rxAudioProcessingModulePtr->gain_control()->set_mode( @@ -5308,7 +5308,7 @@ Channel::GetRemoteRTCPData( { _engineStatisticsPtr->SetLastError( VE_RTP_RTCP_MODULE_ERROR, kTraceError, - "GetRemoteRTCPData() failed to retrieve sender info for remote" + "GetRemoteRTCPData() failed to retrieve sender info for remote " "side"); return -1; } @@ -5354,7 +5354,7 @@ Channel::GetRemoteRTCPData( WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId), "GetRemoteRTCPData() failed to measure statistics due" - "to lack of received RTP and/or RTCP packets"); + " to lack of received RTP and/or RTCP packets"); } if (NULL != jitter) { @@ -5450,7 +5450,7 @@ Channel::GetRTPStatistics( { _engineStatisticsPtr->SetLastError( VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning, - "GetRTPStatistics() failed to read RTP statistics from the" + "GetRTPStatistics() failed to read RTP statistics from the " "RTP/RTCP module"); } @@ -5468,7 +5468,7 @@ Channel::GetRTPStatistics( WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId), "GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu," - "discardedPackets = %lu)", + " discardedPackets = %lu)", averageJitterMs, maxJitterMs, discardedPackets); return 0; } @@ -5506,7 +5506,7 @@ Channel::GetRTPStatistics(CallStatistics& stats) WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId), "GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu," - "extendedMax=%lu, jitterSamples=%li)", + " extendedMax=%lu, jitterSamples=%li)", stats.fractionLost, stats.cumulativeLost, stats.extendedMax, stats.jitterSamples); @@ -5518,7 +5518,7 @@ Channel::GetRTPStatistics(CallStatistics& stats) { WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId), - "GetRTPStatistics() RTCP is disabled => valid RTT" + "GetRTPStatistics() RTCP is disabled => valid RTT " "measurements cannot be retrieved"); } else { @@ -5535,14 +5535,14 @@ Channel::GetRTPStatistics(CallStatistics& stats) { WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId), - "GetRTPStatistics() failed to retrieve RTT from" + "GetRTPStatistics() failed to retrieve RTT from " "the RTP/RTCP module"); } } else { WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId), - "GetRTPStatistics() failed to measure RTT since no" + "GetRTPStatistics() failed to measure RTT since no " "RTP packets have been received yet"); } } @@ -5568,7 +5568,7 @@ Channel::GetRTPStatistics(CallStatistics& stats) WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId), "GetRTPStatistics() failed to retrieve RTP datacounters =>" - "output will not be complete"); + " output will not be complete"); } stats.bytesSent = bytesSent; @@ -5579,7 +5579,7 @@ Channel::GetRTPStatistics(CallStatistics& stats) WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId), "GetRTPStatistics() => bytesSent=%d, packetsSent=%d," - "bytesReceived=%d, packetsReceived=%d)", + " bytesReceived=%d, packetsReceived=%d)", stats.bytesSent, stats.packetsSent, stats.bytesReceived, stats.packetsReceived); @@ -6652,7 +6652,7 @@ Channel::RegisterReceiveCodecsToRTPModule() kTraceVoice, VoEId(_instanceId, _channelId), "Channel::RegisterReceiveCodecsToRTPModule() %s " - "(%d/%d/%d/%d) has been added to the RTP/RTCP" + "(%d/%d/%d/%d) has been added to the RTP/RTCP " "receiver", codec.plname, codec.pltype, codec.plfreq, codec.channels, codec.rate);