commit c6cf9020349590026f28fa85d9ee92f0ff6e371c
author henrika <henrika@webrtc.org> Wed Jul 29 15:20:13 2020
committer Henrik Andreassson <henrika@webrtc.org> Thu Jul 30 08:10:03 2020
tree ed47131387048a39b3483d7ad9d2d9b43df24390
parent 3ea3e0c345936911ad7c1ff706355dc95e2132bc

Improves logging in MediaChannel

This CL changes the style of logging for an API which is essential when
WebRTC is used in Chrome. By changing the format, we can more easily
tie in (search for tags etc.) logs from WebRTC with logs in Chrome.
See e.g.
https://chromium-review.googlesource.com/c/chromium/src/+/2093443
for more details.

I decided to use a new private method to avoid using rtc::StringBuilder.
The idea was to make the log statements less complex and more condensed.

Tbr: mbonadei
Bug: webrtc:11493
Change-Id: I46b4a933ad62ac1db376743b4a41b62c5f8c6ac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172841
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31808}

media/engine/webrtc_voice_engine.cc

diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 38dc346..016c00f 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -11,7 +11,6 @@
 #include "media/engine/webrtc_voice_engine.h"
 
 #include <algorithm>
-#include <cstdio>
 #include <functional>
 #include <string>
 #include <utility>
@@ -43,6 +42,7 @@
 #include "rtc_base/race_checker.h"
 #include "rtc_base/strings/audio_format_to_string.h"
 #include "rtc_base/strings/string_builder.h"
+#include "rtc_base/strings/string_format.h"
 #include "rtc_base/third_party/base64/base64.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
@@ -2051,14 +2051,19 @@
 
 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
   RTC_DCHECK(worker_thread_checker_.IsCurrent());
+  RTC_LOG(LS_INFO) << rtc::StringFormat("WRVMC::%s({ssrc=%u}, {volume=%.2f})",
+                                        __func__, ssrc, volume);
   const auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
-    RTC_LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc;
+    RTC_LOG(LS_WARNING) << rtc::StringFormat(
+        "WRVMC::%s => (WARNING: no receive stream for SSRC %u)", __func__,
+        ssrc);
     return false;
   }
   it->second->SetOutputVolume(volume);
-  RTC_LOG(LS_INFO) << "SetOutputVolume() to " << volume
-                   << " for recv stream with ssrc " << ssrc;
+  RTC_LOG(LS_INFO) << rtc::StringFormat(
+      "WRVMC::%s => (stream with SSRC %u now uses volume %.2f)", __func__, ssrc,
+      volume);
   return true;
 }
 

pc/remote_audio_source.cc

diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index 301cd3f..18a4ed2 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -22,6 +22,7 @@
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/string_format.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_checker.h"
 
@@ -102,6 +103,8 @@
 void RemoteAudioSource::SetVolume(double volume) {
   RTC_DCHECK_GE(volume, 0);
   RTC_DCHECK_LE(volume, 10);
+  RTC_LOG(LS_INFO) << rtc::StringFormat("RAS::%s({volume=%.2f})", __func__,
+                                        volume);
   for (auto* observer : audio_observers_) {
     observer->OnSetVolume(volume);
   }