| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| D | filemediaengine_unittest.cc | 386 const uint32 send_ssrc = RtpTestUtility::kDefaultSsrc + 1; in TEST_F() local
387 voice_channel_->AddSendStream(StreamParams::CreateLegacy(send_ssrc)); in TEST_F()
403 2 * RtpTestUtility::GetTestPacketCount(), &net_dump, send_ssrc)); in TEST_F()
415 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/app/webrtc/ |
| D | webrtcsession_unittest.cc | 2571 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() local
2572 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
2578 session_->SetAudioSend(send_ssrc, false, options, renderer.get()); in TEST_F()
2579 EXPECT_TRUE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
2585 session_->SetAudioSend(send_ssrc, true, options, NULL); in TEST_F()
2586 EXPECT_FALSE(channel->IsStreamMuted(send_ssrc)); in TEST_F()
2601 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() local
2605 session_->SetAudioSend(send_ssrc, true, options, renderer.get()); in TEST_F()
2614 session_->SetAudioSend(send_ssrc, true, options, NULL); in TEST_F()
2641 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() local
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| D | datachannel.cc | 264 void DataChannel::SetSendSsrc(uint32 send_ssrc) { in SetSendSsrc() argument
269 send_ssrc_ = send_ssrc; in SetSendSsrc()
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| D | datachannel.h | 163 void SetSendSsrc(uint32 send_ssrc);
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| D | webrtcsession.cc | 1070 uint32 send_ssrc = 0; in CanInsertDtmf() local
1074 &send_ssrc)) { in CanInsertDtmf()
1088 uint32 send_ssrc = 0; in InsertDtmf() local
1090 track_id, &send_ssrc))) { in InsertDtmf()
1094 if (!voice_channel_->InsertDtmf(send_ssrc, code, duration, in InsertDtmf()
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| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| D | fakewebrtcvoiceengine.h | 202 send_ssrc(0), in Channel()
233 uint32 send_ssrc; member
292 if (local_ssrc == iter->second->send_ssrc) in GetChannelFromLocalSsrc()
893 channels_[channel]->send_ssrc = ssrc;
898 ssrc = channels_[channel]->send_ssrc;
949 block.source_SSRC = channels_[channel]->send_ssrc;
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| D | webrtcvoiceengine_unittest.cc | 2280 unsigned int send_ssrc; in TEST_F() local
2281 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num, send_ssrc)); in TEST_F()
2282 EXPECT_NE(0U, send_ssrc); in TEST_F()
2283 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num, send_ssrc)); in TEST_F()
2284 EXPECT_EQ(kSsrc1, send_ssrc); in TEST_F()
2340 unsigned int send_ssrc; in TEST_F() local
2341 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num1, send_ssrc)); in TEST_F()
2342 EXPECT_EQ(kSsrc1, send_ssrc); in TEST_F()
2346 EXPECT_EQ(0, voe_.GetLocalSSRC(channel_num2, send_ssrc)); in TEST_F()
2347 EXPECT_EQ(kSsrc1, send_ssrc); in TEST_F()
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| D | webrtcvideoengine.h | 292 uint32 send_ssrc() const { return 0; } in send_ssrc() function
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| D | webrtcvoiceengine.cc | 2703 unsigned int send_ssrc = 0; in SetSendCodecs() local
2705 if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) { in SetSendCodecs()
2709 if (rtp->SetLocalSSRC(channel, send_ssrc) == -1) { in SetSendCodecs()
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| /external/chromium_org/third_party/webrtc/video/ |
| D | video_send_stream_tests.cc | 230 FakeReceiveStatistics(uint32_t send_ssrc, in FakeReceiveStatistics() argument
237 stats_map_[send_ssrc] = lossy_stats_.get(); in FakeReceiveStatistics()
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