| 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,
|
|
|