(Auto)update libjingle 69587333-> 69588608
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/media/webrtc/webrtcexport.h b/media/webrtc/webrtcexport.h
index 71ebe4e..b00b0cb 100644
--- a/media/webrtc/webrtcexport.h
+++ b/media/webrtc/webrtcexport.h
@@ -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
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 7df4788..9f6673a 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -1002,6 +1002,9 @@
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 @@
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;
}
diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h
index efc388f..2e50dee 100644
--- a/media/webrtc/webrtcvoiceengine.h
+++ b/media/webrtc/webrtcvoiceengine.h
@@ -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
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index 58893b9..ad44757 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -3176,4 +3176,3 @@
CoUninitialize();
}
#endif
-
