| /external/chromium_org/media/cast/video_sender/ |
| video_sender.cc | 48 video_config.rtp_config.ssrc,
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| video_sender_unittest.cc | 159 video_config.rtp_config.ssrc = 1;
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| /external/bluetooth/bluedroid/stack/avdt/ |
| avdt_api.c | 1357 UINT32 ssrc; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/H264/ |
| rtp_sender_h264.cc | 221 const uint32_t ssrc)
243 ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc);
318 const uint32_t ssrc,
347 // make a copy and only change the SSRC and seqNum
365 //ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, ssrc);
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvideoengine_unittest.cc | 120 void OnMediaError(uint32 ssrc, cricket::VideoMediaChannel::Error error) {
1600 unsigned int ssrc = 0; local
1621 unsigned int ssrc = 0; local
[all...] |
| webrtcvideoengine2_unittest.cc | 789 recv_stream->GetConfig().rtp.rtx.begin()->second.ssrc);
801 recv_stream->GetConfig().rtp.rtx.begin()->second.ssrc);
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| peerconnectioninterface_unittest.cc | 94 // Gets the first ssrc of given content type from the ContentInfo.
95 bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
96 if (!content_info || !ssrc) {
105 *ssrc = media_desc->streams().begin()->first_ssrc();
110 const char kSdpSsrcAtribute[] = "a=ssrc:";
111 const char kSdpSsrcAtributeZero[] = "a=ssrc:0";
549 // that start with a:ssrc"
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_receiver_unittest.cc | 71 rtp_header_.header.ssrc = 0x12345678; // Arbitrary.
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| audio_coding_module_unittest.cc | 51 rtp_header->header.ssrc = 0x1234;
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| initial_delay_manager_unittest.cc | 38 rtp_info->header.ssrc = 0x87654321; // Arbitrary.
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| Channel.cc | 30 rtpInfo.header.ssrc = 0;
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| TestAllCodecs.cc | 63 rtp_info.header.ssrc = 0;
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_sender.cc | 329 const uint32_t* SSRC)
344 _rembSSRC[i] = SSRC[i];
400 RTCPSender::SetSSRC( const uint32_t ssrc)
411 _SSRC = ssrc;
414 void RTCPSender::SetRemoteSSRC(uint32_t ssrc)
417 _remoteSSRC = ssrc;
452 int32_t RTCPSender::AddMixedCNAME(const uint32_t SSRC,
462 _csrcCNAMEs[SSRC] = ptr;
466 int32_t RTCPSender::RemoveMixedCNAME(const uint32_t SSRC) {
469 _csrcCNAMEs.find(SSRC);
[all...] |
| rtp_fec_unittest.cc | 705 // The sequence number, marker bit, and ssrc number are defined in the
710 // The ssrc value for FEC packets is set to the one used for the
712 received_packet->ssrc = ssrc_;
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| rtcp_receiver.cc | 128 RTCPReceiver::SetRemoteSSRC( const uint32_t ssrc)
132 // new SSRC reset old reports
137 _remoteSSRC = ssrc;
178 LOG(LS_WARNING) << "Failed to reset rtt for ssrc " << remoteSSRC;
430 "ssrc", main_ssrc_);
461 "ssrc", main_ssrc_);
489 // |rtcpPacket.ReportBlockItem.SSRC| is the SSRC identifier of the source to
493 if (registered_ssrcs_.find(rtcpPacket.ReportBlockItem.SSRC) ==
517 reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC;
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| normal_test.cc | 110 rtpInfo.header.ssrc = 0;
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| generic_codec_test.cc | 555 rtpInfo.header.ssrc = 0;
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| /external/chromium_org/content/browser/resources/media/ |
| webrtc_internals.js | 127 * Extracts ssrc info from a setLocal/setRemoteDescription update.
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| /external/chromium_org/media/cast/test/ |
| sender.cc | 103 audio_config.rtp_config.ssrc = 1;
136 video_config.rtp_config.ssrc = 11;
[all...] |
| /external/compiler-rt/lib/dfsan/ |
| dfsan_custom.cc | 214 dfsan_label *sdest = shadow_for(dest), *ssrc = shadow_for((void *)src); local
215 internal_memcpy((void *)sdest, (void *)ssrc, n * sizeof(dfsan_label));
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| /external/chromium_org/media/cast/rtcp/ |
| rtcp_sender_unittest.cc | 31 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
47 virtual bool SendRtcpPacket(uint32 ssrc,
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| packet_buffer_unittest.cc | 53 packet->header.ssrc = 0x12345678;
|
| /external/webrtc/src/modules/interface/ |
| module_common_types.h | 23 WebRtc_UWord32 ssrc; member in struct:webrtc::RTPHeader
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| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| srtpfilter_unittest.cc | 740 void OnSrtpError(uint32 ssrc, cricket::SrtpFilter::Mode mode,
742 ssrc_ = ssrc;
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
| vie_autotest_rtp_rtcp.cc | 34 const unsigned int SSRC)
120 // Set an SSRC to avoid issues with collisions.
445 // Test to set SSRC
451 ViETest::Log("Set SSRC %u", setSSRC);
460 ViETest::Log("Received SSRC %u\n", receivedSSRC);
756 // SSRC
763 unsigned int ssrc = 0; local
764 EXPECT_EQ(0, ViE.rtp_rtcp->GetLocalSSRC(tbChannel.videoChannel, ssrc));
[all...] |
