From c6cf9020349590026f28fa85d9ee92f0ff6e371c Mon Sep 17 00:00:00 2001 From: henrika Date: Wed, 29 Jul 2020 17:20:13 +0200 Subject: [PATCH] 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 Reviewed-by: Henrik Andreassson Cr-Commit-Position: refs/heads/master@{#31808} --- media/engine/webrtc_voice_engine.cc | 13 +++++++++---- pc/remote_audio_source.cc | 3 +++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index 38dc3462ac..016c00fbcc 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 -#include #include #include #include @@ -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::SetLocalSource(uint32_t ssrc, 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; } diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc index 301cd3fb5b..18a4ed25c8 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 @@ bool RemoteAudioSource::remote() const { 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); }