From 10bd88e2b52e8175f396cd7b1e6b1f5422c2cd0f Mon Sep 17 00:00:00 2001 From: "henrike@webrtc.org" Date: Tue, 11 Mar 2014 21:07:25 +0000 Subject: [PATCH] (Auto)update libjingle 62871616-> 62948689 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5683 4adac7df-926f-26a2-2b94-8c16560cd09d --- talk/media/base/mediachannel.h | 7 ++++++- talk/media/webrtc/webrtcvideoengine.cc | 9 +++++++-- talk/p2p/base/session.cc | 2 ++ 3 files changed, 15 insertions(+), 3 deletions(-) diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h index 0af14a961c..1a7d0c93c4 100644 --- a/talk/media/base/mediachannel.h +++ b/talk/media/base/mediachannel.h @@ -319,6 +319,7 @@ struct VideoOptions { dscp.SetFrom(change.dscp); suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate); unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit); + use_simulcast_adapter.SetFrom(change.use_simulcast_adapter); } bool operator==(const VideoOptions& o) const { @@ -345,7 +346,8 @@ struct VideoOptions { lower_min_bitrate == o.lower_min_bitrate && dscp == o.dscp && suspend_below_min_bitrate == o.suspend_below_min_bitrate && - unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit; + unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit && + use_simulcast_adapter == o.use_simulcast_adapter; } std::string ToString() const { @@ -377,6 +379,7 @@ struct VideoOptions { suspend_below_min_bitrate); ost << ToStringIfSet("num channels for early receive", unsignalled_recv_stream_limit); + ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter); ost << "}"; return ost.str(); } @@ -428,6 +431,8 @@ struct VideoOptions { Settable suspend_below_min_bitrate; // Limit on the number of early receive channels that can be created. Settable unsignalled_recv_stream_limit; + // Enable use of simulcast adapter. + Settable use_simulcast_adapter; }; // A class for playing out soundclips. diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc index 685b3d50c4..5439bc9143 100644 --- a/talk/media/webrtc/webrtcvideoengine.cc +++ b/talk/media/webrtc/webrtcvideoengine.cc @@ -2818,6 +2818,7 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) { conference_mode_turned_off = true; } + // Save the options, to be interpreted where appropriate. // Use options_.SetAll() instead of assignment so that unset value in options // will not overwrite the previous option value. @@ -2854,9 +2855,12 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) { expected_bitrate = kMaxVideoBitrate; } - if (send_codec_ && + bool reset_send_codec_needed = send_codec_ && (send_max_bitrate_ != expected_bitrate || denoiser_changed || - adjusted_min_bitrate)) { + adjusted_min_bitrate); + + + if (reset_send_codec_needed) { // On success, SetSendCodec() will reset send_max_bitrate_ to // expected_bitrate. if (!SetSendCodec(*send_codec_, @@ -2867,6 +2871,7 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) { } LogSendCodecChange("SetOptions()"); } + if (leaky_bucket_changed) { bool enable_leaky_bucket = options_.video_leaky_bucket.GetWithDefaultIfUnset(false); diff --git a/talk/p2p/base/session.cc b/talk/p2p/base/session.cc index 05ac207091..3520984339 100644 --- a/talk/p2p/base/session.cc +++ b/talk/p2p/base/session.cc @@ -548,6 +548,8 @@ TransportProxy* BaseSession::GetOrCreateTransportProxy( new TransportWrapper(transport)); transproxy->SignalCandidatesReady.connect( this, &BaseSession::OnTransportProxyCandidatesReady); + if (identity_) + transproxy->SetIdentity(identity_); transports_[content_name] = transproxy; return transproxy;