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(Auto)update libjingle 62871616-> 62948689

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5683 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org 2014-03-11 21:07:25 +00:00
parent 21df84711a
commit 10bd88e2b5
3 changed files with 15 additions and 3 deletions
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7
talk/media/base/mediachannel.h
View File

@ -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<bool> suspend_below_min_bitrate;
// Limit on the number of early receive channels that can be created.
Settable<int> unsignalled_recv_stream_limit;
// Enable use of simulcast adapter.
Settable<bool> use_simulcast_adapter;
};
// A class for playing out soundclips.

9
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -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);

2
talk/p2p/base/session.cc
View File

@ -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;