Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository.
TBR=wu@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1824004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index 63a32e8..d6a8b58 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -69,9 +69,9 @@
#include "talk/media/devices/videorendererfactory.h"
#include "talk/media/webrtc/webrtcvideocapturer.h"
#include "third_party/icu/public/common/unicode/unistr.h"
-#include "third_party/webrtc/system_wrappers/interface/trace.h"
-#include "third_party/webrtc/video_engine/include/vie_base.h"
-#include "third_party/webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/video_engine/include/vie_base.h"
+#include "webrtc/voice_engine/include/voe_base.h"
using icu::UnicodeString;
using webrtc::AudioSourceInterface;
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index e56e8a2..78619c6 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -66,8 +66,6 @@
using cricket::NS_GINGLE_P2P;
using cricket::NS_JINGLE_ICE_UDP;
using cricket::TransportInfo;
-using cricket::kDtmfDelay;
-using cricket::kDtmfReset;
using talk_base::SocketAddress;
using talk_base::scoped_ptr;
using webrtc::CreateSessionDescription;
@@ -2247,13 +2245,13 @@
cricket::SessionDescription* answer_copy = answer->description()->Copy();
answer_copy->RemoveContentByName("video");
- JsepSessionDescription* modified_answer =
- new JsepSessionDescription(JsepSessionDescription::kAnswer);
+ talk_base::scoped_ptr<JsepSessionDescription> modified_answer(
+ new JsepSessionDescription(JsepSessionDescription::kAnswer));
EXPECT_TRUE(modified_answer->Initialize(answer_copy,
answer->session_id(),
answer->session_version()));
- SetLocalDescriptionExpectError(kMlineMismatch, modified_answer);
+ SetLocalDescriptionExpectError(kMlineMismatch, modified_answer.get());
SetLocalDescriptionWithoutError(answer);
}
diff --git a/talk/build/common.gypi b/talk/build/common.gypi
index 7775e24..f2322b1 100644
--- a/talk/build/common.gypi
+++ b/talk/build/common.gypi
@@ -29,6 +29,7 @@
{
'variables': {
+ 'webrtc_root%': '<(DEPTH)/webrtc',
# TODO(ronghuawu): Chromium build will need a different libjingle_root.
'libjingle_root%': '<(DEPTH)',
# TODO(ronghuawu): For now, disable the Chrome plugins, which causes a
@@ -44,6 +45,7 @@
'../..',
'../../third_party',
'../../third_party/webrtc',
+ '../../webrtc',
],
'defines': [
'EXPAT_RELATIVE_PATH',
diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp
index b1d7131..270a55d 100755
--- a/talk/libjingle.gyp
+++ b/talk/libjingle.gyp
@@ -80,7 +80,7 @@
{
'variables': {
'java_src_dir': 'app/webrtc/java/src',
- 'webrtc_modules_dir': '<(DEPTH)/third_party/webrtc/modules',
+ 'webrtc_modules_dir': '<(webrtc_root)/modules',
'peerconnection_java_files': [
'app/webrtc/java/src/org/webrtc/AudioSource.java',
'app/webrtc/java/src/org/webrtc/AudioTrack.java',
@@ -105,13 +105,13 @@
# included here, or better yet, build a proper .jar in webrtc
# and include it here.
'android_java_files': [
- '<(webrtc_modules_dir)/audio_device/android/org/webrtc/voiceengine/WebRTCAudioDevice.java',
- '<(webrtc_modules_dir)/video_capture/android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java',
- '<(webrtc_modules_dir)/video_capture/android/java/org/webrtc/videoengine/VideoCaptureAndroid.java',
- '<(webrtc_modules_dir)/video_capture/android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java',
- '<(webrtc_modules_dir)/video_render/android/java/org/webrtc/videoengine/ViEAndroidGLES20.java',
- '<(webrtc_modules_dir)/video_render/android/java/org/webrtc/videoengine/ViERenderer.java',
- '<(webrtc_modules_dir)/video_render/android/java/org/webrtc/videoengine/ViESurfaceRenderer.java',
+ '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java',
+ '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/CaptureCapabilityAndroid.java',
+ '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java',
+ '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java',
+ '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java',
+ '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViERenderer.java',
+ '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java',
],
},
'action_name': 'create_jar',
@@ -769,11 +769,11 @@
'type': 'static_library',
'dependencies': [
'<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
- '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture_module',
- '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render_module',
- '<(DEPTH)/third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
- '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine',
- '<(DEPTH)/third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/modules/modules.gyp:video_capture_module',
+ '<(webrtc_root)/modules/modules.gyp:video_render_module',
+ '<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core',
+ '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'libjingle',
'libjingle_sound',
],
diff --git a/talk/libjingle.scons b/talk/libjingle.scons
index 2bacf6b..cfc0249 100644
--- a/talk/libjingle.scons
+++ b/talk/libjingle.scons
@@ -22,13 +22,13 @@
)
talk.Library(env, name = "gunit",
srcs = [
- "third_party/gtest/src/gtest-all.cc",
+ "testing/gtest/src/gtest-all.cc",
],
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
cppdefines = [
"EXPAT_RELATIVE_PATH",
@@ -355,7 +355,7 @@
includedirs = [
"third_party/libudev",
"third_party/expat-2.0.1/lib",
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/srtp/include",
"third_party/srtp/crypto/include",
] + SSL_INCLUDES,
@@ -405,10 +405,10 @@
"base/unittest_main.cc",
],
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
cppdefines = [
"EXPAT_RELATIVE_PATH",
@@ -589,10 +589,10 @@
"base/windowpicker_unittest.cc",
],
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
win_srcs = [
"base/win32_unittest.cc",
@@ -644,10 +644,10 @@
"p2p/client/portallocator_unittest.cc",
],
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
libs = [
"jingle",
@@ -688,10 +688,10 @@
"session/media/ssrcmuxfilter_unittest.cc",
],
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
libs = [
"jingle",
@@ -712,10 +712,10 @@
mac_libs = SSL_LIBS,
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
cppdefines = [
"EXPAT_RELATIVE_PATH",
@@ -738,10 +738,10 @@
],
mac_libs = SSL_LIBS,
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
cppdefines = [
"EXPAT_RELATIVE_PATH",
@@ -772,10 +772,10 @@
"xmpp/xmppstanzaparser_unittest.cc",
],
includedirs = [
- "third_party/gtest/include",
+ "testing/gtest/include",
"third_party/expat-2.0.1/lib",
"third_party/srtp",
- "third_party/gtest",
+ "testing/gtest",
],
libs = [
"jingle",
diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp
index 5a2f3a9..4b7b793 100755
--- a/talk/libjingle_tests.gyp
+++ b/talk/libjingle_tests.gyp
@@ -225,8 +225,7 @@
['os_posix==1', {
'sources': [
'base/sslidentity_unittest.cc',
- # TODO(ronghuawu): reenable once fixed on build bots.
- # 'base/sslstreamadapter_unittest.cc',
+ 'base/sslstreamadapter_unittest.cc',
],
}],
], # conditions
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index b20051e..441fbff 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -393,12 +393,6 @@
DF_SEND = 0x02,
};
-// Special purpose DTMF event code used by the VoiceMediaChannel::InsertDtmf.
-const int kDtmfDelay = -1; // Insert a delay to the end of the DTMF queue.
-const int kDtmfReset = -2; // Reset the DTMF queue.
-// The delay in ms when the InsertDtmf is called with kDtmfDelay.
-const int kDtmfDelayInMs = 2000;
-
class MediaChannel : public sigslot::has_slots<> {
public:
class NetworkInterface {
@@ -738,10 +732,8 @@
// Send and/or play a DTMF |event| according to the |flags|.
// The DTMF out-of-band signal will be used on sending.
// The |ssrc| should be either 0 or a valid send stream ssrc.
- // The valid value for the |event| are -2 to 15.
- // kDtmfReset(-2) is used to reset the DTMF.
- // kDtmfDelay(-1) is used to insert a delay to the end of the DTMF queue.
- // 0 to 15 which corresponding to DTMF event 0-9, *, #, A-D.
+ // The valid value for the |event| are 0 to 15 which corresponding to
+ // DTMF event 0-9, *, #, A-D.
virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
diff --git a/talk/media/base/rtpdataengine_unittest.cc b/talk/media/base/rtpdataengine_unittest.cc
index 23b57d4..4ee072e 100644
--- a/talk/media/base/rtpdataengine_unittest.cc
+++ b/talk/media/base/rtpdataengine_unittest.cc
@@ -140,7 +140,8 @@
std::string GetSentData(int index) {
// Assume RTP header of length 12
- const talk_base::Buffer* packet = iface_->GetRtpPacket(index);
+ talk_base::scoped_ptr<const talk_base::Buffer> packet(
+ iface_->GetRtpPacket(index));
if (packet->length() > 12) {
return std::string(packet->data() + 12, packet->length() - 12);
} else {
@@ -149,7 +150,8 @@
}
cricket::RtpHeader GetSentDataHeader(int index) {
- const talk_base::Buffer* packet = iface_->GetRtpPacket(index);
+ talk_base::scoped_ptr<const talk_base::Buffer> packet(
+ iface_->GetRtpPacket(index));
cricket::RtpHeader header;
GetRtpHeader(packet->data(), packet->length(), &header);
return header;
diff --git a/talk/media/webrtc/fakewebrtcvideoengine.h b/talk/media/webrtc/fakewebrtcvideoengine.h
index 886f403..68963f7 100644
--- a/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -708,6 +708,10 @@
WEBRTC_STUB(DeregisterDecoderObserver, (const int));
WEBRTC_STUB(SendKeyFrame, (const int));
WEBRTC_STUB(WaitForFirstKeyFrame, (const int, const bool));
+#ifdef USE_WEBRTC_DEV_BRANCH
+ WEBRTC_STUB(StartDebugRecording, (int, const char*));
+ WEBRTC_STUB(StopDebugRecording, (int));
+#endif
// webrtc::ViECapture
WEBRTC_STUB(NumberOfCaptureDevices, ());
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 266cefc..f392751 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2290,12 +2290,6 @@
return false;
}
- // TODO(ronghuawu): Remove this once the reset and delay are supported by VoE.
- // https://code.google.com/p/webrtc/issues/detail?id=747
- if (event == kDtmfReset || event == kDtmfDelay) {
- return true;
- }
-
// Send the event.
if (flags & cricket::DF_SEND) {
if (send_ssrc_ != ssrc && ssrc != 0) {
diff --git a/talk/p2p/base/tcpport.cc b/talk/p2p/base/tcpport.cc
index 356dd67..037abec 100644
--- a/talk/p2p/base/tcpport.cc
+++ b/talk/p2p/base/tcpport.cc
@@ -121,8 +121,11 @@
ICE_TYPE_PREFERENCE_HOST_TCP, true);
} else {
LOG_J(LS_INFO, this) << "Not listening due to firewall restrictions.";
- // Sending error signal as we can't allocate tcp candidate.
- SignalPortError(this);
+ // Note: We still add the address, since otherwise the remote side won't
+ // recognize our incoming TCP connections.
+ AddAddress(talk_base::SocketAddress(ip(), 0),
+ talk_base::SocketAddress(ip(), 0), TCP_PROTOCOL_NAME,
+ LOCAL_PORT_TYPE, ICE_TYPE_PREFERENCE_HOST_TCP, true);
}
}
diff --git a/talk/p2p/client/portallocator_unittest.cc b/talk/p2p/client/portallocator_unittest.cc
index 2113103..1ee97f1 100644
--- a/talk/p2p/client/portallocator_unittest.cc
+++ b/talk/p2p/client/portallocator_unittest.cc
@@ -505,18 +505,6 @@
// There is no error reporting from RelayEntry to handle this failure.
}
-TEST_F(PortAllocatorTest, TestTcpPortNoListenAllowed) {
- AddInterface(kClientAddr);
- allocator().set_flags(cricket::PORTALLOCATOR_DISABLE_UDP |
- cricket::PORTALLOCATOR_DISABLE_STUN |
- cricket::PORTALLOCATOR_DISABLE_RELAY);
- allocator().set_allow_tcp_listen(false);
- EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
- session_->StartGettingPorts();
- EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
- EXPECT_TRUE(candidates_.empty());
-}
-
// Testing STUN timeout.
TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
fss_->AddRule(false, talk_base::FP_UDP, talk_base::FD_ANY, kClientAddr);
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index ddf7c67..b49d5a0 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -385,10 +385,8 @@
// Send and/or play a DTMF |event| according to the |flags|.
// The DTMF out-of-band signal will be used on sending.
// The |ssrc| should be either 0 or a valid send stream ssrc.
- // The valid value for the |event| are -2 to 15.
- // kDtmfReset(-2) is used to reset the DTMF.
- // kDtmfDelay(-1) is used to insert a delay to the end of the DTMF queue.
- // 0 to 15 which corresponding to DTMF event 0-9, *, #, A-D.
+ // The valid value for the |event| are 0 which corresponding to DTMF
+ // event 0-9, *, #, A-D.
bool InsertDtmf(uint32 ssrc, int event_code, int duration, int flags);
bool SetOutputScaling(uint32 ssrc, double left, double right);
// Get statistics about the current media session.
diff --git a/talk/session/media/channel_unittest.cc b/talk/session/media/channel_unittest.cc
index 8c02505..c9918f8 100644
--- a/talk/session/media/channel_unittest.cc
+++ b/talk/session/media/channel_unittest.cc
@@ -57,8 +57,6 @@
using cricket::CA_ANSWER;
using cricket::CA_UPDATE;
using cricket::FakeVoiceMediaChannel;
-using cricket::kDtmfDelay;
-using cricket::kDtmfReset;
using cricket::ScreencastId;
using cricket::StreamParams;
using cricket::TransportChannel;
@@ -2221,23 +2219,17 @@
EXPECT_TRUE(SendAccept());
EXPECT_EQ(0U, media_channel1_->dtmf_info_queue().size());
- EXPECT_TRUE(channel1_->InsertDtmf(-1, kDtmfReset, -1, cricket::DF_SEND));
- EXPECT_TRUE(channel1_->InsertDtmf(0, kDtmfDelay, 90, cricket::DF_PLAY));
EXPECT_TRUE(channel1_->InsertDtmf(1, 3, 100, cricket::DF_SEND));
EXPECT_TRUE(channel1_->InsertDtmf(2, 5, 110, cricket::DF_PLAY));
EXPECT_TRUE(channel1_->InsertDtmf(3, 7, 120,
cricket::DF_PLAY | cricket::DF_SEND));
- ASSERT_EQ(5U, media_channel1_->dtmf_info_queue().size());
+ ASSERT_EQ(3U, media_channel1_->dtmf_info_queue().size());
EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[0],
- -1, kDtmfReset, -1, cricket::DF_SEND));
- EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[1],
- 0, kDtmfDelay, 90, cricket::DF_PLAY));
- EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[2],
1, 3, 100, cricket::DF_SEND));
- EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[3],
+ EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[1],
2, 5, 110, cricket::DF_PLAY));
- EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[4],
+ EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[2],
3, 7, 120, cricket::DF_PLAY | cricket::DF_SEND));
}
