| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdataengine_unittest.cc | 230 params.ssrc = 42;
283 EXPECT_EQ(header0.ssrc, 42U);
295 EXPECT_EQ(header1.ssrc, 42U);
332 params1.ssrc = 41;
334 params2.ssrc = 42;
382 params.ssrc = 42;
415 // PT= 103, SN=2, TS=3, SSRC = 4, data = "abcde"
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| 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.
|
| rtputils.cc | 159 GetRtpSsrc(data, len, &(header->ssrc)));
169 // This method returns SSRC first of RTCP packet, except if packet is SDES.
173 // Packet should be at least of 8 bytes, to get SSRC from a RTCP packet.
223 SetRtpSsrc(data, len, header.ssrc));
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| 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.
|
| /external/chromium/third_party/libjingle/source/talk/session/phone/ |
| rtpdump.cc | 82 bool RtpDumpPacket::GetRtpSsrc(uint32* ssrc) const {
83 if (!ssrc || !IsValidRtpPacket()) {
86 *ssrc = talk_base::GetBE32(&data[8]);
|
| rtpdump.h | 75 // Get the sequence number, timestampe, and SSRC of the RTP packet. Return
79 bool GetRtpSsrc(uint32* ssrc) const;
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| mediaengine.h | 269 virtual bool AddStream(uint32 ssrc) { return true; }
270 virtual bool RemoveStream(uint32 ssrc) { return true; }
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| call.h | 58 void SetVideoRenderer(BaseSession *session, uint32 ssrc,
|
| srtpfilter.cc | 377 policy.ssrc.type = static_cast<ssrc_type_t>(type);
378 policy.ssrc.value = 0;
423 LOG(LS_INFO) << "SRTP event: SSRC collision";
|
| 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...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| mediamessages_unittest.cc | 61 static std::string ViewVideoStaticVgaXml(const std::string& ssrc) {
65 " ssrc='" + ssrc + "'"
117 "<ssrc>" + ssrc1 + "</ssrc>"
118 "<ssrc>" + ssrc2 + "</ssrc>"
119 "<ssrc-group"
122 "<ssrc>" + ssrc1 + "</ssrc>"
[all...] |
| channel_unittest.cc | 256 // Add stream information (SSRC) to the local content but not to the remote
257 // content. This means that we per default know the SSRC of what we send but
262 // If SSRC_MUX is used we also need to know the SSRC of the incoming stream.
293 // Add stream information (SSRC) to the local content but not to the remote
294 // content. This means that we per default know the SSRC of what we send but
299 // If SSRC_MUX is used we also need to know the SSRC of the incoming stream.
404 bool SendCustomRtp1(uint32 ssrc, int sequence_number) {
405 std::string data(CreateRtpData(ssrc, sequence_number));
409 bool SendCustomRtp2(uint32 ssrc, int sequence_number) {
410 std::string data(CreateRtpData(ssrc, sequence_number))
[all...] |
| channelmanager.h | 182 bool RegisterVoiceProcessor(uint32 ssrc,
185 bool UnregisterVoiceProcessor(uint32 ssrc,
|
| mediasessionclient.cc | 279 // Parses an ssrc string as a legacy stream. If it fails, returns
285 uint32 ssrc; local
286 if (!talk_base::FromString(ssrc_str, &ssrc)) {
287 return BadParse("Missing or invalid ssrc.", error);
290 streams->push_back(StreamParams::CreateLegacy(ssrc));
770 buzz::XmlElement* CreateGingleSsrcElem(const buzz::QName& name, uint32 ssrc) {
772 if (ssrc) {
773 SetXmlBody(elem, ssrc);
[all...] |
| /external/chromium_org/content/browser/resources/media/ |
| stats_graph_helper.js | 8 // e.g. 1234-0-ssrc-abcd123-bytesSent is the graph for the series of bytesSent
9 // for ssrc-abcd123 of PeerConnection 0 in process 1234.
85 'ssrc': true,
210 if (report.type == 'ssrc') {
|
| webrtc_internals.css | 17 .ssrc-info-block {
|
| /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
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| rtcp_utility.cc | 343 big_endian_reader.ReadU32(&field_.report_block_item.ssrc);
401 uint32 ssrc; local
403 big_endian_reader.ReadU32(&ssrc);
408 field_.c_name.sender_ssrc = ssrc;
891 big_endian_reader.ReadU32(&field_.fir_item.ssrc);
|
| /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/media/webrtc/ |
| webrtcvoiceengine_unittest.cc | 84 void OnVoiceChannelError(uint32 ssrc,
86 ssrc_ = ssrc;
93 uint32 ssrc() const { function in class:WebRtcVoiceEngineTestFake::ChannelErrorListener
150 void TestInsertDtmf(uint32 ssrc, bool caller) {
168 EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111, cricket::DF_SEND));
175 EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123, cricket::DF_SEND));
180 // Check we fail if the ssrc is invalid.
185 EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123, cricket::DF_SEND));
190 EXPECT_TRUE(channel_->InsertDtmf(ssrc, 3, 134, cricket::DF_PLAY));
196 EXPECT_TRUE(channel_->InsertDtmf(ssrc, 4, 145
2032 unsigned int ssrc = 0; local
2275 unsigned int ssrc = 0; local
[all...] |
| 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...] |
| fakewebrtcvoiceengine.h | 672 WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
674 channels_[channel]->send_ssrc = ssrc;
677 WEBRTC_FUNC(GetLocalSSRC, (int channel, unsigned int& ssrc)) {
679 ssrc = channels_[channel]->send_ssrc;
682 WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
[all...] |
| /external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/ |
| rtp_packetizer.cc | 148 big_endian_writer.WriteU32(config_.ssrc);
|
| /external/chromium_org/third_party/cld/encodings/compact_lang_det/ |
| cldutil.cc | 530 const char* ssrc = reinterpret_cast<const char*>(usrc); local
531 DbgBiTermToStderr(bihash, probs, ssrc, len2);
534 const char* ssrc = reinterpret_cast<const char*>(usrc); local
536 string temp(ssrc, len2);
|
| /external/chromium/third_party/libjingle/source/talk/p2p/base/ |
| sessionmessages.cc | 862 std::string ssrc = source_elem->Attr(QN_GINGLE_NOTIFY_SOURCE_SSRC); local
863 if (ssrc != buzz::STR_EMPTY) {
866 sources.audio_ssrc = strtoul(ssrc.c_str(), NULL, 10);
868 sources.video_ssrc = strtoul(ssrc.c_str(), NULL, 10);
[all...] |
