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Introduce DLOG to video and voiceengine.

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This CL removes a handful of low-importance logging from our release builds.

Bug: webrtc:8529
Change-Id: I1043f501c16ce24a39512307e8cddccf4c4d4ab6
Reviewed-on: https://webrtc-review.googlesource.com/c/47163
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25622}
This commit is contained in:
Jonas Olsson
2018-11-13 14:43:09 +01:00
committed by Commit Bot
parent 318da51f99
commit 8544799cf1
2 changed files with 18 additions and 18 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
media/engine/webrtcvideoengine.cc
View File

@ -791,7 +791,7 @@ webrtc::RTCError WebRtcVideoChannel::SetRtpSendParameters(
// different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
<< "is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
@ -882,7 +882,7 @@ bool WebRtcVideoChannel::SetRtpReceiveParameters(
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) {
RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
<< "unsupported.";
return false;
}
@ -999,7 +999,7 @@ bool WebRtcVideoChannel::SetSend(bool send) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSend");
RTC_LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
if (send && !send_codec_) {
RTC_LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
RTC_DLOG(LS_ERROR) << "SetSend(true) called before setting codec.";
return false;
}
{
@ -2322,7 +2322,7 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetLocalSsrc(
// right now this can't be done due to unittests depending on receiving what
// they are sending from the same MediaChannel.
if (local_ssrc == config_.rtp.local_ssrc) {
RTC_LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
RTC_DLOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
"unchanged; local_ssrc="
<< local_ssrc;
return;

12
media/engine/webrtcvoiceengine.cc
View File

@ -78,11 +78,11 @@ class ProxySink : public webrtc::AudioSinkInterface {
bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) {
RTC_LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
RTC_DLOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
return false;
}
if (sp.ssrcs.size() > 1) {
RTC_LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: "
RTC_DLOG(LS_ERROR) << "Multiple SSRCs in stream parameters: "
<< sp.ToString();
return false;
}
@ -1393,7 +1393,7 @@ webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
// different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
<< "is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER);
}
@ -1491,7 +1491,7 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) {
RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
<< "unsupported.";
return false;
}
@ -1879,7 +1879,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
const uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) {
RTC_LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
RTC_DLOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
return false;
}
@ -2071,7 +2071,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
// Remove oldest unsignaled stream, if we have too many.
if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front();
RTC_LOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
RTC_DLOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
<< remove_ssrc;
RemoveRecvStream(remove_ssrc);
}