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Unified Diff: webrtc/call/call.cc

Issue 1400333002: Log Call {audio, video} stream deletions. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: move logging to ctor/dtor Created 5 years, 2 months ago
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 2c80ccff0dc038b7e23ddafc28f8b598bda3667e..f788b6cbdead51f1047cc51a397ab697362d9945 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -176,16 +176,19 @@ PacketReceiver* Call::Receiver() { return this; }
webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
+ // TODO(pbos): When adding AudioSendStream, add both TRACE_EVENT0 and config
+ // logging to match others.
stefan-webrtc 2015/10/14 10:30:16 Rewrite these comments.
pbos-webrtc 2015/10/15 10:21:42 Done.
return nullptr;
}
void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+ // TODO(pbos): When adding AudioSendStream, add both TRACE_EVENT0 and config
+ // logging to match others.
}
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
- LOG(LS_INFO) << "CreateAudioReceiveStream: " << config.ToString();
AudioReceiveStream* receive_stream = new AudioReceiveStream(
channel_group_->GetRemoteBitrateEstimator(), config);
{
@@ -224,8 +227,6 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
const webrtc::VideoSendStream::Config& config,
const VideoEncoderConfig& encoder_config) {
TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
- LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
- RTC_DCHECK(!config.rtp.ssrcs.empty());
// TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
// the call has already started.
@@ -288,7 +289,6 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
const webrtc::VideoReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
- LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
VideoReceiveStream* receive_stream = new VideoReceiveStream(
num_cpu_cores_, channel_group_.get(),
rtc::AtomicOps::Increment(&next_channel_id_), config,

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