generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

(Auto)update libjingle 69634309-> 69640360

Browse Source 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6512 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org
2014-06-20 19:02:09 +00:00
parent b43c99de29
commit 0d15159b04
4 changed files with 1 additions and 27 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
talk/media/webrtc/webrtcexport.h
View File

@@ -27,9 +27,6 @@
#ifndef TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_ #ifndef TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
#define 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) #if !defined(GOOGLE_CHROME_BUILD) && !defined(CHROMIUM_BUILD)
#define LIBPEERCONNECTION_LIB 1 #define LIBPEERCONNECTION_LIB 1
#endif #endif

20
talk/media/webrtc/webrtcvoiceengine.cc
View File

@@ -1002,9 +1002,6 @@ bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
LOG_RTCERR2(SetRecordingDevice, in_name, in_id); LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
ret = false; ret = false;
} }
webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
if (ap)
ap->Initialize();
} }
// Find the playout device id in VoiceEngine and set playout device. // Find the playout device id in VoiceEngine and set playout device.
@@ -3131,23 +3128,6 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
LOG_RTCERR2(SetInputMute, channel, muted); LOG_RTCERR2(SetInputMute, channel, muted);
return false; 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; return true;
} }

4
talk/media/webrtc/webrtcvoiceengine.h
View File

@@ -47,10 +47,6 @@
#if !defined(LIBPEERCONNECTION_LIB) && \ #if !defined(LIBPEERCONNECTION_LIB) && \
!defined(LIBPEERCONNECTION_IMPLEMENTATION) !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." #error "Bogus include."
#endif #endif

1
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@@ -3176,3 +3176,4 @@ TEST(WebRtcVoiceEngineTest, CoInitialize) {
CoUninitialize(); CoUninitialize();
} }
#endif #endif