| /external/libpcap/ |
| Makefile.in | 78 SSRC = @SSRC@
84 SRC = $(PSRC) $(FSRC) $(CSRC) $(SSRC) $(GENSRC)
88 OBJ = $(PSRC:.c=.o) $(FSRC:.c=.o) $(CSRC:.c=.o) $(SSRC:.c=.o) $(GENSRC:.c=.o) $(LIBOBJS)
|
| /external/srtp/ |
| Changes | 161 Removed 'ssrc' from the srtp_init_aes_128_prf() function argument
163 ssrc which they will be receiving can still use libsrtp. Now the
164 SSRC value is gleaned from the rtp header and exored into the
170 call will be added to the library that enables multiple ssrc/key
|
| /external/chromium/third_party/libjingle/source/talk/session/phone/ |
| rtpdump.h | 75 // Get the sequence number, timestampe, and SSRC of the RTP packet. Return
79 bool GetRtpSsrc(uint32* ssrc) const;
|
| mediasessionclient.cc | 703 // TODO: Figure out how to integrate SSRC into Jingle.
728 // TODO: Figure out how to integrate SSRC into Jingle.
786 buzz::XmlElement* CreateGingleSsrcElem(const buzz::QName& name, uint32 ssrc) {
788 if (ssrc) {
789 SetXmlBody(elem, ssrc);
855 QN_GINGLE_AUDIO_SRCID, audio->ssrc()));
881 QN_GINGLE_VIDEO_SRCID, video->ssrc()));
[all...] |
| rtpdump.cc | 82 bool RtpDumpPacket::GetRtpSsrc(uint32* ssrc) const {
83 if (!ssrc || !IsValidRtpPacket()) {
86 *ssrc = talk_base::GetBE32(&data[8]);
|
| /external/chromium_org/media/cast/net/rtp_sender/ |
| rtp_sender.cc | 30 config_.ssrc = audio_config->sender_ssrc;
38 config_.ssrc = video_config->sender_ssrc;
|
| /external/chromium_org/media/cast/rtcp/ |
| rtcp_utility.h | 50 uint32 ssrc; member in struct:media::cast::RtcpFieldReportBlockItem
90 uint32 ssrc; member in struct:media::cast::RtcpFieldPayloadSpecificFirItem
|
| /external/chromium_org/media/cast/rtp_receiver/rtp_parser/ |
| rtp_parser_unittest.cc | 46 EXPECT_EQ(kTestSsrc, parsed_header.webrtc.header.ssrc);
85 config_.ssrc = kTestSsrc;
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| peerconnection.cc | 692 uint32 ssrc) {
693 stream_handler_container_->AddRemoteAudioTrack(stream, audio_track, ssrc);
698 uint32 ssrc) {
699 stream_handler_container_->AddRemoteVideoTrack(stream, video_track, ssrc);
715 uint32 ssrc) {
716 stream_handler_container_->AddLocalAudioTrack(stream, audio_track, ssrc);
720 uint32 ssrc) {
721 stream_handler_container_->AddLocalVideoTrack(stream, video_track, ssrc);
|
| statstypes.h | 87 // StatsReport of |type| = "ssrc" is statistics for a specific rtp stream.
88 // The |id| field is the SSRC in decimal form of the rtp stream.
|
| datachannel.cc | 194 send_params.ssrc = config_.id;
253 if (params.ssrc != expected_ssrc) {
435 send_params.ssrc = config_.id;
437 send_params.ssrc = send_ssrc_;
|
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdataengine.h | 107 virtual bool RemoveSendStream(uint32 ssrc);
109 virtual bool RemoveRecvStream(uint32 ssrc);
|
| testutils.cc | 67 ret &= buf->ReadUInt32(&ssrc);
78 ssrc == ssc &&
157 size_t count, talk_base::StreamInterface* stream, uint32 ssrc) {
193 ssrc);
349 // There should be an rtx_ssrc per ssrc.
|
| videocommon.h | 38 // TODO(janahan): For now, a hard-coded ssrc is used as the video ssrc.
40 // processing, it doesn't have the correct ssrc. Since currently only Tx
|
| filemediaengine_unittest.cc | 191 uint32 ssrc; local
192 if (!packet.GetRtpSsrc(&ssrc)) {
195 ssrcs.insert(ssrc);
384 // Test that we can specify the ssrc for outgoing RTP packets.
|
| rtputils.h | 43 uint32 ssrc; member in struct:cricket::RtpHeader
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| typingmonitor.cc | 63 void TypingMonitor::OnVoiceChannelError(uint32 ssrc,
73 // multiple sending audio streams. SSRC 0 means the default sending audio
|
| currentspeakermonitor.h | 65 // SSRC. This only fires after the audio monitor on the underlying Call has
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| fakewebrtcvideoengine.h | 301 ssrcs_[0] = 0; // default ssrc.
395 // ssrcs_[0] is the default local ssrc.
900 const unsigned int ssrc,
906 channels_[channel]->ssrcs_[idx] = ssrc;
909 channels_[channel]->rtx_ssrcs_[idx] = ssrc;
918 const webrtc::StreamType usage, const unsigned int ssrc)) {
921 channels_.find(channel)->second->remote_rtx_ssrc_ = ssrc;
928 unsigned int& ssrc))
[all...] |
| /external/srtp/googlepatches/ |
| vidyo-3-srtp-ws.patch | 621 /* set ssrc to that provided */
623 p->ssrc.value);
660 policy.ssrc.type = ssrc_any_inbound;
800 policy.ssrc.value = ssrc;
819 policy.ssrc.type = ssrc_specific;
820 policy.ssrc.value = 0xdecafbad;
853 stream->ssrc,
866 policy.ssrc.type = ssrc_specific;
867 policy.ssrc.value = 0xcafebabe
[all...] |
| /external/chromium_org/content/browser/media/ |
| webrtc_internals_browsertest.cc | 23 ss << "a=ssrc:" << id;
113 // report id (e.g. ssrc-1234) for each stats name (e.g. framerate).
319 // Get the JSON string of the ssrc info from the page.
476 stats.values["ssrc"] = ssrc1.id;
477 ExecuteAndVerifyAddStats(pc_2, "ssrc", "dummyId", stats);
489 const string type = "ssrc";
490 const string id = "ssrc-1234";
|
| /external/chromium_org/third_party/libjingle/source/talk/examples/call/ |
| callclient.h | 98 // Maintain a mapping of (session, ssrc) to rendered view.
266 uint32 ssrc, int width, int height, int framerate,
268 bool RemoveStaticRenderedView(uint32 ssrc);
|
| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| constants.cc | 111 const buzz::StaticQName QN_SSRC = { NS_EMPTY, "ssrc" };
248 const buzz::StaticQName QN_JINGLE_DRAFT_SSRC = { NS_JINGLE_DRAFT, "ssrc" };
250 { NS_JINGLE_DRAFT, "ssrc-group" };
|
| /external/chromium_org/media/cast/ |
| cast_receiver_impl.cc | 116 VLOG(1) << "Received a packet with a non matching sender SSRC "
|
| /external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/ |
| rtp_packetizer.cc | 148 big_endian_writer.WriteU32(config_.ssrc);
|
