| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| D | filemediaengine_unittest.cc | 389 const uint32 send_ssrc = RtpTestUtility::kDefaultSsrc + 1; in TEST_F() local
390 voice_channel_->AddSendStream(StreamParams::CreateLegacy(send_ssrc)); in TEST_F()
406 2 * RtpTestUtility::GetTestPacketCount(), &net_dump, send_ssrc)); in TEST_F()
418 send_ssrc)); in TEST_F()
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| D | rtpdump_unittest.cc | 141 const uint32 send_ssrc = kTestSsrc + 1; in TEST() local
142 reader_w_ssrc.SetSsrc(send_ssrc); in TEST()
149 EXPECT_EQ(send_ssrc, ssrc); in TEST()
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| D | fakemediaengine.h | 170 const uint32 send_ssrc() { in send_ssrc() function
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| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| D | fakewebrtcvoiceengine.h | 100 send_ssrc(0), in Channel()
125 uint32 send_ssrc; member
180 if (local_ssrc == iter->second->send_ssrc) in GetChannelFromLocalSsrc()
674 channels_[channel]->send_ssrc = ssrc;
679 ssrc = channels_[channel]->send_ssrc;
719 block.source_SSRC = channels_[channel]->send_ssrc;
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| D | webrtcvoiceengine_unittest.cc | 1964 unsigned int send_ssrc; in TEST_F() local
1965 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num, send_ssrc)); in TEST_F()
1966 EXPECT_NE(0U, send_ssrc); in TEST_F()
1967 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num, send_ssrc)); in TEST_F()
1968 EXPECT_EQ(kSsrc1, send_ssrc); in TEST_F()
2009 unsigned int send_ssrc; in TEST_F() local
2010 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num1, send_ssrc)); in TEST_F()
2011 EXPECT_EQ(kSsrc1, send_ssrc); in TEST_F()
2015 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num2, send_ssrc)); in TEST_F()
2016 EXPECT_EQ(kSsrc1, send_ssrc); in TEST_F()
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| D | webrtcvideoengine.h | 289 uint32 send_ssrc() const { return 0; } in send_ssrc() function
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| D | webrtcvoiceengine.cc | 2418 unsigned int send_ssrc; in ConfigureRecvChannel() local
2420 if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) { in ConfigureRecvChannel()
2421 LOG_RTCERR2(GetSendSSRC, channel, send_ssrc); in ConfigureRecvChannel()
2424 if (rtp->SetLocalSSRC(channel, send_ssrc) == -1) { in ConfigureRecvChannel()
2425 LOG_RTCERR2(SetSendSSRC, channel, send_ssrc); in ConfigureRecvChannel()
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| D | webrtcvideoengine.cc | 3117 unsigned int send_ssrc = 0; in ConfigureReceiving() local
3119 if (rtp->GetLocalSSRC(vie_channel_, send_ssrc) == -1) { in ConfigureReceiving()
3120 LOG_RTCERR2(GetLocalSSRC, vie_channel_, send_ssrc); in ConfigureReceiving()
3123 if (rtp->SetLocalSSRC(channel_id, send_ssrc) == -1) { in ConfigureReceiving()
3124 LOG_RTCERR2(SetLocalSSRC, channel_id, send_ssrc); in ConfigureReceiving()
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| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| D | webrtcsession_unittest.cc | 2115 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() local
2116 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
2122 session_->SetAudioSend(send_ssrc, false, options, renderer.get()); in TEST_F()
2123 EXPECT_TRUE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
2127 session_->SetAudioSend(send_ssrc, true, options, NULL); in TEST_F()
2128 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
2159 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() local
2160 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
2162 session_->SetVideoSend(send_ssrc, false, options); in TEST_F()
2163 EXPECT_TRUE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
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| D | datachannel.cc | 223 void DataChannel::SetSendSsrc(uint32 send_ssrc) { in SetSendSsrc() argument
228 send_ssrc_ = send_ssrc; in SetSendSsrc()
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| D | datachannel.h | 143 void SetSendSsrc(uint32 send_ssrc);
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| D | webrtcsession.cc | 933 uint32 send_ssrc = 0; in CanInsertDtmf() local
937 &send_ssrc)) { in CanInsertDtmf()
951 uint32 send_ssrc = 0; in InsertDtmf() local
953 track_id, &send_ssrc))) { in InsertDtmf()
957 if (!voice_channel_->InsertDtmf(send_ssrc, code, duration, in InsertDtmf()
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