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Reduce some logging at INFO level by moving log statements

Browse source 
from LS_INFO to LS_VERBOSE.

By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.

Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
This commit is contained in:
Harald Alvestrand 2019-12-12 13:40:50 +01:00 committed by Commit Bot
parent 08c46adc1e
commit 977b265702
7 changed files with 23 additions and 15 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
call/call.cc
View file

@ -1044,8 +1044,15 @@ void Call::UpdateAggregateNetworkState() {
((have_video && video_network_state_ == kNetworkUp) ||
(have_audio && audio_network_state_ == kNetworkUp));
RTC_LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
<< (aggregate_network_up ? "up" : "down");
if (aggregate_network_up != aggregate_network_up_) {
RTC_LOG(LS_INFO)
<< "UpdateAggregateNetworkState: aggregate_state change to "
<< (aggregate_network_up ? "up" : "down");
} else {
RTC_LOG(LS_VERBOSE)
<< "UpdateAggregateNetworkState: aggregate_state remains at "
<< (aggregate_network_up ? "up" : "down");
}
aggregate_network_up_ = aggregate_network_up;
transport_send_ptr_->OnNetworkAvailability(aggregate_network_up);

4
call/rtp_transport_controller_send.cc
View file

@ -302,8 +302,8 @@ void RtpTransportControllerSend::OnNetworkRouteChanged(
}
}
void RtpTransportControllerSend::OnNetworkAvailability(bool network_available) {
RTC_LOG(LS_INFO) << "SignalNetworkState "
<< (network_available ? "Up" : "Down");
RTC_LOG(LS_VERBOSE) << "SignalNetworkState "
<< (network_available ? "Up" : "Down");
NetworkAvailability msg;
msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
msg.network_available = network_available;

8
media/engine/webrtc_voice_engine.cc
View file

@ -233,16 +233,16 @@ void WebRtcVoiceEngine::Init() {
"rtc-low-prio", webrtc::TaskQueueFactory::Priority::LOW)));
// Load our audio codec lists.
RTC_LOG(LS_INFO) << "Supported send codecs in order of preference:";
RTC_LOG(LS_VERBOSE) << "Supported send codecs in order of preference:";
send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders());
for (const AudioCodec& codec : send_codecs_) {
RTC_LOG(LS_INFO) << ToString(codec);
RTC_LOG(LS_VERBOSE) << ToString(codec);
}
RTC_LOG(LS_INFO) << "Supported recv codecs in order of preference:";
RTC_LOG(LS_VERBOSE) << "Supported recv codecs in order of preference:";
recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders());
for (const AudioCodec& codec : recv_codecs_) {
RTC_LOG(LS_INFO) << ToString(codec);
RTC_LOG(LS_VERBOSE) << ToString(codec);
}
#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)

9
modules/audio_processing/aec3/matched_filter.cc
View file

@ -447,10 +447,11 @@ void MatchedFilter::LogFilterProperties(int sample_rate_hz,
int start = static_cast<int>(alignment_shift * downsampling_factor);
int end = static_cast<int>((alignment_shift + filters_[k].size()) *
downsampling_factor);
RTC_LOG(LS_INFO) << "Filter " << k << ": start: "
<< (start - static_cast<int>(shift)) / kFsBy1000
<< " ms, end: "
<< (end - static_cast<int>(shift)) / kFsBy1000 << " ms.";
RTC_LOG(LS_VERBOSE) << "Filter " << k << ": start: "
<< (start - static_cast<int>(shift)) / kFsBy1000
<< " ms, end: "
<< (end - static_cast<int>(shift)) / kFsBy1000
<< " ms.";
alignment_shift += filter_intra_lag_shift_;
}
}

2
pc/data_channel.cc
View file

@ -722,7 +722,7 @@ bool DataChannel::SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) {
cricket::SendDataResult send_result = cricket::SDR_SUCCESS;
bool retval = provider_->SendData(send_params, buffer, &send_result);
if (retval) {
RTC_LOG(LS_INFO) << "Sent CONTROL message on channel " << config_.id;
RTC_LOG(LS_VERBOSE) << "Sent CONTROL message on channel " << config_.id;
if (handshake_state_ == kHandshakeShouldSendAck) {
handshake_state_ = kHandshakeReady;

2
pc/peer_connection.cc
View file

@ -5924,7 +5924,7 @@ cricket::ChannelInterface* PeerConnection::GetChannel(
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
RTC_LOG(LS_VERBOSE)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the SCTP transport.";
return false;

2
rtc_base/openssl_adapter.cc
View file

@ -780,7 +780,7 @@ void OpenSSLAdapter::SSLInfoCallback(const SSL* s, int where, int ret) {
str = "SSL_accept";
}
if (where & SSL_CB_LOOP) {
RTC_DLOG(LS_INFO) << str << ":" << SSL_state_string_long(s);
RTC_DLOG(LS_VERBOSE) << str << ":" << SSL_state_string_long(s);
} else if (where & SSL_CB_ALERT) {
str = (where & SSL_CB_READ) ? "read" : "write";
RTC_DLOG(LS_INFO) << "SSL3 alert " << str << ":"