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(Auto)update libjingle 69587333-> 69588608

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-06-20 00:13:01 +00:00
parent 1ef789d455
commit 8563ef448a
4 changed files with 27 additions and 1 deletions
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3
talk/media/webrtc/webrtcexport.h
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@ -27,6 +27,9 @@
#ifndef TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
#define TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
// When building for Chrome a part of the code can be built into
// a shared library, which is controlled by these macros.
// For all other builds, we always build a static library.
#if !defined(GOOGLE_CHROME_BUILD) && !defined(CHROMIUM_BUILD)
#define LIBPEERCONNECTION_LIB 1
#endif

20
talk/media/webrtc/webrtcvoiceengine.cc
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@ -1002,6 +1002,9 @@ bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
ret = false;
}
webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
if (ap)
ap->Initialize();
}
// Find the playout device id in VoiceEngine and set playout device.
@ -3128,6 +3131,23 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
LOG_RTCERR2(SetInputMute, channel, muted);
return false;
}
// We set the AGC to mute state only when all the channels are muted.
// This implementation is not ideal, instead we should signal the AGC when
// the mic channel is muted/unmuted. We can't do it today because there
// is no good way to know which stream is mapping to the mic channel.
bool all_muted = muted;
for (ChannelMap::const_iterator iter = send_channels_.begin();
iter != send_channels_.end() && all_muted; ++iter) {
if (engine()->voe()->volume()->GetInputMute(iter->second->channel(),
all_muted)) {
LOG_RTCERR1(GetInputMute, iter->second->channel());
return false;
}
}
webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
if (ap)
ap->set_output_will_be_muted(all_muted);
return true;
}

4
talk/media/webrtc/webrtcvoiceengine.h
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@ -47,6 +47,10 @@
#if !defined(LIBPEERCONNECTION_LIB) && \
!defined(LIBPEERCONNECTION_IMPLEMENTATION)
// If you hit this, then you've tried to include this header from outside
// the shared library. An instance of this class must only be created from
// within the library that actually implements it. Otherwise use the
// WebRtcMediaEngine to construct an instance.
#error "Bogus include."
#endif

1
talk/media/webrtc/webrtcvoiceengine_unittest.cc
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@ -3176,4 +3176,3 @@ TEST(WebRtcVoiceEngineTest, CoInitialize) {
CoUninitialize();
}
#endif