From 8544799cf1779d365556a81fb5bc6a1f7cda5577 Mon Sep 17 00:00:00 2001
From: Jonas Olsson <jonasolsson@webrtc.org>
Date: Tue, 13 Nov 2018 14:43:09 +0100
Subject: [PATCH] Introduce DLOG to video and voiceengine.

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}
---
 media/engine/webrtcvideoengine.cc | 16 ++++++++--------
 media/engine/webrtcvoiceengine.cc | 20 ++++++++++----------
 2 files changed, 18 insertions(+), 18 deletions(-)

diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index 74aa55be00..9e54b2b368 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -791,8 +791,8 @@ 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 "
-                      << "is not currently supported.";
+    RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
+                       << "is not currently supported.";
     return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
   }
 
@@ -882,8 +882,8 @@ bool WebRtcVideoChannel::SetRtpReceiveParameters(
 
   webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
   if (current_parameters != parameters) {
-    RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
-                      << "unsupported.";
+    RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
+                       << "unsupported.";
     return false;
   }
   return true;
@@ -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;
   }
   {
@@ -2322,9 +2322,9 @@ 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 "
-                        "unchanged; local_ssrc="
-                     << local_ssrc;
+    RTC_DLOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
+                         "unchanged; local_ssrc="
+                      << local_ssrc;
     return;
   }
 
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index eaf8746aee..5c96668a4d 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -78,12 +78,12 @@ 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: "
-                      << sp.ToString();
+    RTC_DLOG(LS_ERROR) << "Multiple SSRCs in stream parameters: "
+                       << sp.ToString();
     return false;
   }
   return true;
@@ -1393,8 +1393,8 @@ 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 "
-                      << "is not currently supported.";
+    RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
+                       << "is not currently supported.";
     return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER);
   }
 
@@ -1491,8 +1491,8 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
 
   webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
   if (current_parameters != parameters) {
-    RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
-                      << "unsupported.";
+    RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
+                       << "unsupported.";
     return false;
   }
   return true;
@@ -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;
   }
 
@@ -2071,8 +2071,8 @@ 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="
-                     << remove_ssrc;
+    RTC_DLOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
+                      << remove_ssrc;
     RemoveRecvStream(remove_ssrc);
   }
   RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());