Modify the gips wrapper for GIPS release 4.0.
Change-Id: I882bfcc02e3968ce012890a8a15be28310973a1b
diff --git a/talk/libjingle.scons b/talk/libjingle.scons
index 6713149..5ad357e 100644
--- a/talk/libjingle.scons
+++ b/talk/libjingle.scons
@@ -119,6 +119,7 @@
"SRTP_RELATIVE_PATH",
"XML_STATIC",
"HAVE_GIPS",
+ "BRUNO",
],
srcs = [
"base/asyncfile.cc",
diff --git a/talk/session/phone/gips.h b/talk/session/phone/gips.h
index 14589fd..2c02271 100644
--- a/talk/session/phone/gips.h
+++ b/talk/session/phone/gips.h
@@ -5,19 +5,18 @@
#define TALK_SESSION_PHONE_GIPS_H_
#include "talk/base/common.h"
-#include "talk/third_party/gips/Interface/GIPSVEBase.h"
-#include "talk/third_party/gips/Interface/GIPSVECodec.h"
-#include "talk/third_party/gips/Interface/GIPSVEDTMF.h"
-#include "talk/third_party/gips/Interface/GIPSVEErrors.h"
-#include "talk/third_party/gips/Interface/GIPSVEExternalMedia.h"
-#include "talk/third_party/gips/Interface/GIPSVEFile.h"
-#include "talk/third_party/gips/Interface/GIPSVEHardware.h"
-#include "talk/third_party/gips/Interface/GIPSVENetwork.h"
-#include "talk/third_party/gips/Interface/GIPSVENetEqStats.h"
-#include "talk/third_party/gips/Interface/GIPSVERTP_RTCP.h"
-#include "talk/third_party/gips/Interface/GIPSVEVideoSync.h"
-#include "talk/third_party/gips/Interface/GIPSVEVolumeControl.h"
-#include "talk/third_party/gips/Interface/GIPSVEVQE.h"
+#include "gips/Interface/GIPSVEBase.h"
+#include "gips/Interface/GIPSVECodec.h"
+#include "gips/Interface/GIPSVEDTMF.h"
+#include "gips/Interface/GIPSVEErrors.h"
+#include "gips/Interface/GIPSVEFile.h"
+#include "gips/Interface/GIPSVEHardware.h"
+#include "gips/Interface/GIPSVENetwork.h"
+#include "gips/Interface/GIPSVENetEqStats.h"
+#include "gips/Interface/GIPSVERTP_RTCP.h"
+#include "gips/Interface/GIPSVEVideoSync.h"
+#include "gips/Interface/GIPSVEVolumeControl.h"
+#include "gips/Interface/GIPSVEVQE.h"
// Tracing helpers, for easy logging when GIPS calls fail.
// Example: "LOG_GIPSERR1(StartSend, channel);" produces the trace
@@ -80,7 +79,6 @@
dtmf_(engine_),
file_(engine_),
hw_(engine_),
- media_(engine_),
neteq_(engine_),
network_(engine_),
rtp_(engine_),
@@ -89,8 +87,7 @@
vqe_(engine_) {
}
GipsWrapper(GIPSVEBase* base, GIPSVECodec* codec, GIPSVEDTMF* dtmf,
- GIPSVEFile* file, GIPSVEHardware* hw,
- GIPSVEExternalMedia* media, GIPSVENetEqStats* neteq,
+ GIPSVEFile* file, GIPSVEHardware* hw, GIPSVENetEqStats* neteq,
GIPSVENetwork* network, GIPSVERTP_RTCP* rtp,
GIPSVEVideoSync* sync, GIPSVEVolumeControl* volume,
GIPSVEVQE* vqe)
@@ -100,7 +97,6 @@
dtmf_(dtmf),
file_(file),
hw_(hw),
- media_(media),
neteq_(neteq),
network_(network),
rtp_(rtp),
@@ -114,7 +110,6 @@
GIPSVEDTMF* dtmf() { return dtmf_.get(); }
GIPSVEFile* file() { return file_.get(); }
GIPSVEHardware* hw() { return hw_.get(); }
- GIPSVEExternalMedia* media() { return media_.get(); }
GIPSVENetEqStats* neteq() { return neteq_.get(); }
GIPSVENetwork* network() { return network_.get(); }
GIPSVERTP_RTCP* rtp() { return rtp_.get(); }
@@ -130,7 +125,6 @@
scoped_gips_ptr<GIPSVEDTMF> dtmf_;
scoped_gips_ptr<GIPSVEFile> file_;
scoped_gips_ptr<GIPSVEHardware> hw_;
- scoped_gips_ptr<GIPSVEExternalMedia> media_;
scoped_gips_ptr<GIPSVENetEqStats> neteq_;
scoped_gips_ptr<GIPSVENetwork> network_;
scoped_gips_ptr<GIPSVERTP_RTCP> rtp_;
diff --git a/talk/session/phone/gipsmediaengine.cc b/talk/session/phone/gipsmediaengine.cc
index 4f8efa2..3cba73e 100644
--- a/talk/session/phone/gipsmediaengine.cc
+++ b/talk/session/phone/gipsmediaengine.cc
@@ -982,20 +982,6 @@
return true;
}
-bool GipsVoiceEngine::RegisterProcessor(
- VoiceProcessor* voice_processor,
- MediaProcessorDirection direction) {
- // stubbing out...
- return true;
-}
-
-bool GipsVoiceEngine::UnregisterProcessor(
- VoiceProcessor* voice_processor,
- MediaProcessorDirection direction) {
- // stubbing out...
- return true;
-}
-
// GipsVoiceMediaChannel
GipsVoiceMediaChannel::GipsVoiceMediaChannel(GipsVoiceEngine *engine)
: GipsMediaChannel<VoiceMediaChannel, GipsVoiceEngine>(
@@ -1332,11 +1318,14 @@
// Tandberg-bridged conferences have an AGC target that is lower than
// GTV-only levels.
+ /* :BRUNO: disable this since this is built for tandberg */
+#ifndef BRUNO
if ((channel_options_ & OPT_AGC_TANDBERG_LEVELS) && !agc_adjusted_) {
if (engine()->AdjustAgcLevel(kTandbergDbAdjustment)) {
agc_adjusted_ = true;
}
}
+#endif
// GIPS resets sequence number when StopSend is called. This
// sometimes causes libSRTP to complain about packets being
diff --git a/talk/session/phone/gipsmediaengine.h b/talk/session/phone/gipsmediaengine.h
index 86cc3f5..2796768 100644
--- a/talk/session/phone/gipsmediaengine.h
+++ b/talk/session/phone/gipsmediaengine.h
@@ -39,7 +39,6 @@
#include "talk/base/scoped_ptr.h"
#include "talk/base/stream.h"
#include "talk/session/phone/channel.h"
-#include "talk/session/phone/mediaprocessorinterface.h"
#include "talk/session/phone/mediaengine.h"
#include "talk/session/phone/gips.h"
#include "talk/session/phone/rtputils.h"
@@ -127,11 +126,6 @@
GipsWrapper* gips_sc() { return gips_sc_.get(); }
int GetLastGipsError();
- virtual bool RegisterVoiceProcessor(VoiceProcessor* vp_interface,
- MediaProcessorDirection direction);
- virtual bool UnregisterVoiceProcessor(VoiceProcessor* vp_interface,
- MediaProcessorDirection direction);
-
private:
typedef std::vector<GipsSoundclipMedia *> SoundclipList;
typedef std::vector<GipsVoiceMediaChannel *> ChannelList;
@@ -181,10 +175,6 @@
GIPS_AGC_config default_agc_config_;
bool initialized_;
- sigslot::signal2<uint32, AudioFrame*> SignalTxMediaFrame;
- sigslot::signal2<uint32, bool> SignalTxMute;
- sigslot::signal2<uint32, AudioFrame*> SignalRxMediaFrame;
- sigslot::signal2<uint32, bool> SignalRxMute;
talk_base::CriticalSection signal_media_critical_;
};
diff --git a/talk/session/phone/mediaengine.cc b/talk/session/phone/mediaengine.cc
index 51e3b8b..992fc15 100644
--- a/talk/session/phone/mediaengine.cc
+++ b/talk/session/phone/mediaengine.cc
@@ -36,6 +36,9 @@
#if defined(HAVE_WEBRTC_VIDEO)
#include "talk/session/phone/webrtcvideoengine.h"
#endif // HAVE_WEBRTC_VIDEO
+#if defined(HAVE_GIPS)
+#include "talk/session/phone/gipsmediaengine.h"
+#endif // HAVE_GIPS
#endif // HAVE_LINPHONE
namespace cricket {
@@ -47,6 +50,11 @@
#define VIDEO_ENG_NAME WebRtcVideoEngine
#endif
+#if defined(HAVE_GIPS)
+#define AUDIO_ENG_NAME GipsVoiceEngine
+#define VIDEO_ENG_NAME NullVideoEngine
+#endif
+
MediaEngineInterface* MediaEngineFactory::Create() {
#if defined(HAVE_LINPHONE)
return new LinphoneMediaEngine("", "");
