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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

16
media/engine/webrtcvideoengine.cc
View File

@ -791,8 +791,8 @@ webrtc::RTCError WebRtcVideoChannel::SetRtpSendParameters(
// different order (which should change the send codec). // different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc); webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) { 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."; << "is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR); return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
} }
@ -882,8 +882,8 @@ bool WebRtcVideoChannel::SetRtpReceiveParameters(
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc); webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) { 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."; << "unsupported.";
return false; return false;
} }
return true; return true;
@ -999,7 +999,7 @@ bool WebRtcVideoChannel::SetSend(bool send) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSend"); TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSend");
RTC_LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false"); RTC_LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
if (send && !send_codec_) { 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; return false;
} }
{ {
@ -2322,9 +2322,9 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetLocalSsrc(
// right now this can't be done due to unittests depending on receiving what // right now this can't be done due to unittests depending on receiving what
// they are sending from the same MediaChannel. // they are sending from the same MediaChannel.
if (local_ssrc == config_.rtp.local_ssrc) { 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=" "unchanged; local_ssrc="
<< local_ssrc; << local_ssrc;
return; return;
} }

20
media/engine/webrtcvoiceengine.cc
View File

@ -78,12 +78,12 @@ class ProxySink : public webrtc::AudioSinkInterface {
bool ValidateStreamParams(const StreamParams& sp) { bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) { 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; return false;
} }
if (sp.ssrcs.size() > 1) { 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(); << sp.ToString();
return false; return false;
} }
return true; return true;
@ -1393,8 +1393,8 @@ webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
// different order (which should change the send codec). // different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc); webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) { 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."; << "is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER); return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER);
} }
@ -1491,8 +1491,8 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc); webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) { 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."; << "unsupported.";
return false; return false;
} }
return true; return true;
@ -1879,7 +1879,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
const uint32_t ssrc = sp.first_ssrc(); const uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) { 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; return false;
} }
@ -2071,8 +2071,8 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
// Remove oldest unsignaled stream, if we have too many. // Remove oldest unsignaled stream, if we have too many.
if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) { if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front(); 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; << remove_ssrc;
RemoveRecvStream(remove_ssrc); RemoveRecvStream(remove_ssrc);
} }
RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size()); RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());