| Index: content/renderer/media/rtc_peer_connection_handler.cc
|
| diff --git a/content/renderer/media/rtc_peer_connection_handler.cc b/content/renderer/media/rtc_peer_connection_handler.cc
|
| index 6b12cd114bd4f93f728e432140268deb66da2f77..f5aec5e191627aa153d2c99b63cd9887ab8352e2 100644
|
| --- a/content/renderer/media/rtc_peer_connection_handler.cc
|
| +++ b/content/renderer/media/rtc_peer_connection_handler.cc
|
| @@ -301,14 +301,13 @@ void CopyConstraintsIntoRtcConfiguration(
|
| &configuration->enable_rtp_data_channel)) {
|
| configuration->enable_rtp_data_channel = false;
|
| }
|
| - // TODO: Special treatment for screencast min bitrate, since it's an integer.
|
| - // if (FindConstraint(constraints,
|
| - // MediaConstraintsInterface::kScreencastMinBitrate,
|
| - // &configuration->screencast_min_bitrate, NULL)) {
|
| - // configuration->override_screencast_min_bitrate = true;
|
| - // }
|
| - // Note: If an optional is not present, webrtc decides on its own
|
| - // what the value should be.
|
| + int rate;
|
| + if (GetConstraintValueAsInteger(
|
| + constraints,
|
| + &blink::WebMediaTrackConstraintSet::googScreencastMinBitrate,
|
| + &rate)) {
|
| + configuration->screencast_min_bitrate = rtc::Optional<int>(rate);
|
| + }
|
| configuration->combined_audio_video_bwe = ConstraintToOptional(
|
| constraints,
|
| &blink::WebMediaTrackConstraintSet::googCombinedAudioVideoBwe);
|
|
|