Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index 41b2c83d382a9be2f8d21adea1f550cf6eaf6131..0ccfb611e4229157f5e3da441386ace83f90c6a8 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 AudioSendStream constructor. |
return nullptr; |
} |
void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { |
+ // TODO(pbos): When adding AudioSendStream, add both TRACE_EVENT0 and config |
+ // logging to AudioSendStream destructor. |
} |
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, |