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Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository.

Browse Source 
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
This commit is contained in:
henrike@webrtc.org 2013-07-17 14:42:53 +00:00
parent a3f30143b7
commit 9de257d00f
14 changed files with 60 additions and 88 deletions
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6
talk/app/webrtc/java/jni/peerconnection_jni.cc
View File

@ -69,9 +69,9 @@
#include "talk/media/devices/videorendererfactory.h" #include "talk/media/devices/videorendererfactory.h"
#include "talk/media/webrtc/webrtcvideocapturer.h" #include "talk/media/webrtc/webrtcvideocapturer.h"
#include "third_party/icu/public/common/unicode/unistr.h" #include "third_party/icu/public/common/unicode/unistr.h"
#include "third_party/webrtc/system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace.h"
#include "third_party/webrtc/video_engine/include/vie_base.h" #include "webrtc/video_engine/include/vie_base.h"
#include "third_party/webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_base.h"
using icu::UnicodeString; using icu::UnicodeString;
using webrtc::AudioSourceInterface; using webrtc::AudioSourceInterface;

8
talk/app/webrtc/webrtcsession_unittest.cc
View File

@ -66,8 +66,6 @@ using cricket::FakeVoiceMediaChannel;
using cricket::NS_GINGLE_P2P; using cricket::NS_GINGLE_P2P;
using cricket::NS_JINGLE_ICE_UDP; using cricket::NS_JINGLE_ICE_UDP;
using cricket::TransportInfo; using cricket::TransportInfo;
using cricket::kDtmfDelay;
using cricket::kDtmfReset;
using talk_base::SocketAddress; using talk_base::SocketAddress;
using talk_base::scoped_ptr; using talk_base::scoped_ptr;
using webrtc::CreateSessionDescription; using webrtc::CreateSessionDescription;
@ -2247,13 +2245,13 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
cricket::SessionDescription* answer_copy = answer->description()->Copy(); cricket::SessionDescription* answer_copy = answer->description()->Copy();
answer_copy->RemoveContentByName("video"); answer_copy->RemoveContentByName("video");
JsepSessionDescription* modified_answer = talk_base::scoped_ptr<JsepSessionDescription> modified_answer(
new JsepSessionDescription(JsepSessionDescription::kAnswer); new JsepSessionDescription(JsepSessionDescription::kAnswer));
EXPECT_TRUE(modified_answer->Initialize(answer_copy, EXPECT_TRUE(modified_answer->Initialize(answer_copy,
answer->session_id(), answer->session_id(),
answer->session_version())); answer->session_version()));
SetLocalDescriptionExpectError(kMlineMismatch, modified_answer); SetLocalDescriptionExpectError(kMlineMismatch, modified_answer.get());
SetLocalDescriptionWithoutError(answer); SetLocalDescriptionWithoutError(answer);
} }

2
talk/build/common.gypi
View File

@ -29,6 +29,7 @@
{ {
'variables': { 'variables': {
'webrtc_root%': '<(DEPTH)/webrtc',
# TODO(ronghuawu): Chromium build will need a different libjingle_root. # TODO(ronghuawu): Chromium build will need a different libjingle_root.
'libjingle_root%': '<(DEPTH)', 'libjingle_root%': '<(DEPTH)',
# TODO(ronghuawu): For now, disable the Chrome plugins, which causes a # TODO(ronghuawu): For now, disable the Chrome plugins, which causes a
@ -44,6 +45,7 @@
'../..', '../..',
'../../third_party', '../../third_party',
'../../third_party/webrtc', '../../third_party/webrtc',
'../../webrtc',
], ],
'defines': [ 'defines': [
'EXPAT_RELATIVE_PATH', 'EXPAT_RELATIVE_PATH',

26
talk/libjingle.gyp
View File

@ -80,7 +80,7 @@
{ {
'variables': { 'variables': {
'java_src_dir': 'app/webrtc/java/src', 'java_src_dir': 'app/webrtc/java/src',
'webrtc_modules_dir': '<(DEPTH)/third_party/webrtc/modules', 'webrtc_modules_dir': '<(webrtc_root)/modules',
'peerconnection_java_files': [ 'peerconnection_java_files': [
'app/webrtc/java/src/org/webrtc/AudioSource.java', 'app/webrtc/java/src/org/webrtc/AudioSource.java',
'app/webrtc/java/src/org/webrtc/AudioTrack.java', 'app/webrtc/java/src/org/webrtc/AudioTrack.java',
@ -105,13 +105,13 @@
# included here, or better yet, build a proper .jar in webrtc # included here, or better yet, build a proper .jar in webrtc
# and include it here. # and include it here.
'android_java_files': [ 'android_java_files': [
'<(webrtc_modules_dir)/audio_device/android/org/webrtc/voiceengine/WebRTCAudioDevice.java', '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java',
'<(webrtc_modules_dir)/video_capture/android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java', '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/CaptureCapabilityAndroid.java',
'<(webrtc_modules_dir)/video_capture/android/java/org/webrtc/videoengine/VideoCaptureAndroid.java', '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java',
'<(webrtc_modules_dir)/video_capture/android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java', '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java',
'<(webrtc_modules_dir)/video_render/android/java/org/webrtc/videoengine/ViEAndroidGLES20.java', '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java',
'<(webrtc_modules_dir)/video_render/android/java/org/webrtc/videoengine/ViERenderer.java', '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViERenderer.java',
'<(webrtc_modules_dir)/video_render/android/java/org/webrtc/videoengine/ViESurfaceRenderer.java', '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java',
], ],
}, },
'action_name': 'create_jar', 'action_name': 'create_jar',
@ -769,11 +769,11 @@
'type': 'static_library', 'type': 'static_library',
'dependencies': [ 'dependencies': [
'<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv', '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
'<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture_module', '<(webrtc_root)/modules/modules.gyp:video_capture_module',
'<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render_module', '<(webrtc_root)/modules/modules.gyp:video_render_module',
'<(DEPTH)/third_party/webrtc/video_engine/video_engine.gyp:video_engine_core', '<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core',
'<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine', '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(DEPTH)/third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers', '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'libjingle', 'libjingle',
'libjingle_sound', 'libjingle_sound',
], ],

36
talk/libjingle.scons
View File

@ -22,13 +22,13 @@ talk.Library(env, name = "expat",
) )
talk.Library(env, name = "gunit", talk.Library(env, name = "gunit",
srcs = [ srcs = [
"third_party/gtest/src/gtest-all.cc", "testing/gtest/src/gtest-all.cc",
], ],
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
cppdefines = [ cppdefines = [
"EXPAT_RELATIVE_PATH", "EXPAT_RELATIVE_PATH",
@ -355,7 +355,7 @@ talk.Library(env, name = "jingle",
includedirs = [ includedirs = [
"third_party/libudev", "third_party/libudev",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/gtest/include", "testing/gtest/include",
"third_party/srtp/include", "third_party/srtp/include",
"third_party/srtp/crypto/include", "third_party/srtp/crypto/include",
] + SSL_INCLUDES, ] + SSL_INCLUDES,
@ -405,10 +405,10 @@ talk.Library(env, name = "unittest_main",
"base/unittest_main.cc", "base/unittest_main.cc",
], ],
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
cppdefines = [ cppdefines = [
"EXPAT_RELATIVE_PATH", "EXPAT_RELATIVE_PATH",
@ -589,10 +589,10 @@ talk.Unittest(env, name = "base",
"base/windowpicker_unittest.cc", "base/windowpicker_unittest.cc",
], ],
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
win_srcs = [ win_srcs = [
"base/win32_unittest.cc", "base/win32_unittest.cc",
@ -644,10 +644,10 @@ talk.Unittest(env, name = "p2p",
"p2p/client/portallocator_unittest.cc", "p2p/client/portallocator_unittest.cc",
], ],
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
libs = [ libs = [
"jingle", "jingle",
@ -688,10 +688,10 @@ talk.Unittest(env, name = "media",
"session/media/ssrcmuxfilter_unittest.cc", "session/media/ssrcmuxfilter_unittest.cc",
], ],
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
libs = [ libs = [
"jingle", "jingle",
@ -712,10 +712,10 @@ talk.Unittest(env, name = "sound",
mac_libs = SSL_LIBS, mac_libs = SSL_LIBS,
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
cppdefines = [ cppdefines = [
"EXPAT_RELATIVE_PATH", "EXPAT_RELATIVE_PATH",
@ -738,10 +738,10 @@ talk.Unittest(env, name = "xmllite",
], ],
mac_libs = SSL_LIBS, mac_libs = SSL_LIBS,
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
cppdefines = [ cppdefines = [
"EXPAT_RELATIVE_PATH", "EXPAT_RELATIVE_PATH",
@ -772,10 +772,10 @@ talk.Unittest(env, name = "xmpp",
"xmpp/xmppstanzaparser_unittest.cc", "xmpp/xmppstanzaparser_unittest.cc",
], ],
includedirs = [ includedirs = [
"third_party/gtest/include", "testing/gtest/include",
"third_party/expat-2.0.1/lib", "third_party/expat-2.0.1/lib",
"third_party/srtp", "third_party/srtp",
"third_party/gtest", "testing/gtest",
], ],
libs = [ libs = [
"jingle", "jingle",

3
talk/libjingle_tests.gyp
View File

@ -225,8 +225,7 @@
['os_posix==1', { ['os_posix==1', {
'sources': [ 'sources': [
'base/sslidentity_unittest.cc', 'base/sslidentity_unittest.cc',
# TODO(ronghuawu): reenable once fixed on build bots. 'base/sslstreamadapter_unittest.cc',
# 'base/sslstreamadapter_unittest.cc',
], ],
}], }],
], # conditions ], # conditions

12
talk/media/base/mediachannel.h
View File

@ -393,12 +393,6 @@ enum DtmfFlags {
DF_SEND = 0x02, 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<> { class MediaChannel : public sigslot::has_slots<> {
public: public:
class NetworkInterface { class NetworkInterface {
@ -738,10 +732,8 @@ class VoiceMediaChannel : public MediaChannel {
// Send and/or play a DTMF |event| according to the |flags|. // Send and/or play a DTMF |event| according to the |flags|.
// The DTMF out-of-band signal will be used on sending. // The DTMF out-of-band signal will be used on sending.
// The |ssrc| should be either 0 or a valid send stream ssrc. // The |ssrc| should be either 0 or a valid send stream ssrc.
// The valid value for the |event| are -2 to 15. // The valid value for the |event| are 0 to 15 which corresponding to
// kDtmfReset(-2) is used to reset the DTMF. // DTMF event 0-9, *, #, A-D.
// 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.
virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) = 0; virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) = 0;
// Gets quality stats for the channel. // Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0; virtual bool GetStats(VoiceMediaInfo* info) = 0;

6
talk/media/base/rtpdataengine_unittest.cc
View File

@ -140,7 +140,8 @@ class RtpDataMediaChannelTest : public testing::Test {
std::string GetSentData(int index) { std::string GetSentData(int index) {
// Assume RTP header of length 12 // 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) { if (packet->length() > 12) {
return std::string(packet->data() + 12, packet->length() - 12); return std::string(packet->data() + 12, packet->length() - 12);
} else { } else {
@ -149,7 +150,8 @@ class RtpDataMediaChannelTest : public testing::Test {
} }
cricket::RtpHeader GetSentDataHeader(int index) { 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; cricket::RtpHeader header;
GetRtpHeader(packet->data(), packet->length(), &header); GetRtpHeader(packet->data(), packet->length(), &header);
return header; return header;

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

@ -708,6 +708,10 @@ class FakeWebRtcVideoEngine
WEBRTC_STUB(DeregisterDecoderObserver, (const int)); WEBRTC_STUB(DeregisterDecoderObserver, (const int));
WEBRTC_STUB(SendKeyFrame, (const int)); WEBRTC_STUB(SendKeyFrame, (const int));
WEBRTC_STUB(WaitForFirstKeyFrame, (const int, const bool)); 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::ViECapture
WEBRTC_STUB(NumberOfCaptureDevices, ()); WEBRTC_STUB(NumberOfCaptureDevices, ());

6
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -2290,12 +2290,6 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event,
return false; 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. // Send the event.
if (flags & cricket::DF_SEND) { if (flags & cricket::DF_SEND) {
if (send_ssrc_ != ssrc && ssrc != 0) { if (send_ssrc_ != ssrc && ssrc != 0) {

7
talk/p2p/base/tcpport.cc
View File

@ -121,8 +121,11 @@ void TCPPort::PrepareAddress() {
ICE_TYPE_PREFERENCE_HOST_TCP, true); ICE_TYPE_PREFERENCE_HOST_TCP, true);
} else { } else {
LOG_J(LS_INFO, this) << "Not listening due to firewall restrictions."; LOG_J(LS_INFO, this) << "Not listening due to firewall restrictions.";
// Sending error signal as we can't allocate tcp candidate. // Note: We still add the address, since otherwise the remote side won't
SignalPortError(this); // 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);
} }
} }

12
talk/p2p/client/portallocator_unittest.cc
View File

@ -505,18 +505,6 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoSockets) {
// There is no error reporting from RelayEntry to handle this failure. // 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. // Testing STUN timeout.
TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpAllowed) { TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
fss_->AddRule(false, talk_base::FP_UDP, talk_base::FD_ANY, kClientAddr); fss_->AddRule(false, talk_base::FP_UDP, talk_base::FD_ANY, kClientAddr);

6
talk/session/media/channel.h
View File

@ -385,10 +385,8 @@ class VoiceChannel : public BaseChannel {
// Send and/or play a DTMF |event| according to the |flags|. // Send and/or play a DTMF |event| according to the |flags|.
// The DTMF out-of-band signal will be used on sending. // The DTMF out-of-band signal will be used on sending.
// The |ssrc| should be either 0 or a valid send stream ssrc. // The |ssrc| should be either 0 or a valid send stream ssrc.
// The valid value for the |event| are -2 to 15. // The valid value for the |event| are 0 which corresponding to DTMF
// kDtmfReset(-2) is used to reset the DTMF. // event 0-9, *, #, A-D.
// 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.
bool InsertDtmf(uint32 ssrc, int event_code, int duration, int flags); bool InsertDtmf(uint32 ssrc, int event_code, int duration, int flags);
bool SetOutputScaling(uint32 ssrc, double left, double right); bool SetOutputScaling(uint32 ssrc, double left, double right);
// Get statistics about the current media session. // Get statistics about the current media session.

14
talk/session/media/channel_unittest.cc
View File

@ -57,8 +57,6 @@ using cricket::CA_PRANSWER;
using cricket::CA_ANSWER; using cricket::CA_ANSWER;
using cricket::CA_UPDATE; using cricket::CA_UPDATE;
using cricket::FakeVoiceMediaChannel; using cricket::FakeVoiceMediaChannel;
using cricket::kDtmfDelay;
using cricket::kDtmfReset;
using cricket::ScreencastId; using cricket::ScreencastId;
using cricket::StreamParams; using cricket::StreamParams;
using cricket::TransportChannel; using cricket::TransportChannel;
@ -2221,23 +2219,17 @@ TEST_F(VoiceChannelTest, TestInsertDtmf) {
EXPECT_TRUE(SendAccept()); EXPECT_TRUE(SendAccept());
EXPECT_EQ(0U, media_channel1_->dtmf_info_queue().size()); 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(1, 3, 100, cricket::DF_SEND));
EXPECT_TRUE(channel1_->InsertDtmf(2, 5, 110, cricket::DF_PLAY)); EXPECT_TRUE(channel1_->InsertDtmf(2, 5, 110, cricket::DF_PLAY));
EXPECT_TRUE(channel1_->InsertDtmf(3, 7, 120, EXPECT_TRUE(channel1_->InsertDtmf(3, 7, 120,
cricket::DF_PLAY | cricket::DF_SEND)); 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], 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)); 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)); 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)); 3, 7, 120, cricket::DF_PLAY | cricket::DF_SEND));
} }