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Review URL: http://webrtc-codereview.appspot.com/148002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@602 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org 2011-09-15 20:49:50 +00:00
parent 4537c2a464
commit fcd12b3b7d
1 changed files with 18 additions and 18 deletions
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36
src/voice_engine/main/source/channel.cc
View File

@ -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);