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(Auto)update libjingle 72205295-> 72320533

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-07-31 15:08:53 +00:00
parent e1c9caf6ee
commit 6b21b71068
4 changed files with 27 additions and 19 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_ #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
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@ -1009,6 +1009,9 @@ 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.
@ -3136,6 +3139,23 @@ 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
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@ -47,6 +47,10 @@
#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

19
talk/media/webrtc/webrtcvoiceengine_unittest.cc
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@ -3185,22 +3185,3 @@ TEST(WebRtcVoiceEngineTest, CoInitialize) {
CoUninitialize(); CoUninitialize();
} }
#endif #endif
#ifdef USE_WEBRTC_DEV_BRANCH
TEST_F(WebRtcVoiceEngineTestFake, ExperimentalNsConfigViaOptions) {
EXPECT_TRUE(SetupEngine());
cricket::FakeAudioProcessing* audio_processing =
static_cast<cricket::FakeAudioProcessing*>(
engine_.voe()->base()->audio_processing());
EXPECT_FALSE(audio_processing->experimental_ns_enabled());
cricket::AudioOptions options;
options.experimental_ns.Set(true);
EXPECT_TRUE(engine_.SetOptions(options));
EXPECT_TRUE(audio_processing->experimental_ns_enabled());
}
#endif