generic-library/webrtc
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(Auto)update libjingle 74696326-> 74723281

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7047 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-09-03 21:50:32 +00:00
parent 1b8b4c4959
commit 609f987488
2 changed files with 11 additions and 2 deletions
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2
talk/media/webrtc/webrtcmediaengine.cc
View File

@ -50,7 +50,6 @@ class WebRtcMediaEngine :
WebRtcVideoDecoderFactory* decoder_factory) {
voice_.SetAudioDeviceModule(adm, adm_sc);
video_.SetVoiceEngine(&voice_);
video_.EnableTimedRender();
video_.SetExternalEncoderFactory(encoder_factory);
video_.SetExternalDecoderFactory(decoder_factory);
}
@ -66,7 +65,6 @@ class WebRtcMediaEngine2 :
WebRtcVideoDecoderFactory* decoder_factory) {
voice_.SetAudioDeviceModule(adm, adm_sc);
video_.SetVoiceEngine(&voice_);
video_.EnableTimedRender();
}
};
#endif // WEBRTC_CHROMIUM_BUILD

11
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -81,6 +81,9 @@ const int kMaxVideoBitrate = 2000;
const int kCpuMonitorPeriodMs = 2000; // 2 seconds.
// TODO(pthatcher): Figure out what the proper value here is, or if we
// can just remove this altogether.
static const int kDefaultRenderDelayMs = 100;
static const int kDefaultLogSeverity = rtc::LS_WARNING;
@ -951,6 +954,9 @@ void WebRtcVideoEngine::Construct(ViEWrapper* vie_wrapper,
LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
}
// Consider jitter, packet loss, etc when rendering. This will
// theoretically make rendering more smooth.
EnableTimedRender();
// Load our RTP Header extensions.
rtp_header_extensions_.push_back(
@ -3392,6 +3398,11 @@ bool WebRtcVideoMediaChannel::ConfigureReceiving(int channel_id,
return false;
}
if (engine()->vie()->render()->SetExpectedRenderDelay(
channel_id, kDefaultRenderDelayMs)) {
LOG_RTCERR2(SetExpectedRenderDelay,
channel_id, kDefaultRenderDelayMs);
}
if (engine_->vie()->rtp()->SetRembStatus(channel_id,
kNotSending,