| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| voiceprocessor.h | 50 virtual void OnFrame(uint32 ssrc,
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| 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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| rtpdump.cc | 91 bool RtpDumpPacket::GetRtpSsrc(uint32* ssrc) const {
93 cricket::GetRtpSsrc(&data[0], data.size(), ssrc);
110 void RtpDumpReader::SetSsrc(uint32 ssrc) {
111 ssrc_override_ = ssrc;
151 // If the packet is RTP and we have specified a ssrc, replace the RTP ssrc
152 // with the specified ssrc.
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| videoprocessor.h | 46 virtual void OnFrame(uint32 ssrc, VideoFrame* frame, bool* drop_frame) = 0;
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| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| 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...] |
| call.h | 78 void SetVideoRenderer(Session* session, uint32 ssrc,
95 const std::string& stream_name, uint32 ssrc,
98 const std::string& stream_name, uint32 ssrc);
192 void OnSpeakerMonitor(CurrentSpeakerMonitor* monitor, uint32 ssrc);
227 bool StopScreencastWithoutSendingUpdate(Session* session, uint32 ssrc);
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| mediasession.h | 233 // Legacy streams have an ssrc, but nothing else.
234 void AddLegacyStream(uint32 ssrc) {
235 streams_.push_back(StreamParams::CreateLegacy(ssrc));
237 void AddLegacyStream(uint32 ssrc, uint32 fid_ssrc) {
238 StreamParams sp = StreamParams::CreateLegacy(ssrc);
239 sp.AddFidSsrc(ssrc, fid_ssrc);
475 uint32 ssrc, StreamParams* stream_out);
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| typingmonitor.h | 71 void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error);
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| /external/srtp/googlepatches/ |
| google-9-rdbx-leak-plug.patch | 265 srtp_remove_stream(srtp_t session, uint32_t ssrc) {
276 while ((stream != NULL) && (ssrc != stream->ssrc)) {
277 @@ -1352,8 +1324,20 @@ srtp_remove_stream(srtp_t session, uint32_t ssrc) {
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| vidyo-4-srtp-rtx.patch | 37 /* set ssrc to that provided */
38 str->ssrc = ssrc;
69 policy.ssrc.type = ssrc_any_inbound;
133 stream->ssrc,
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| /frameworks/av/media/libstagefright/timedtext/ |
| TimedTextPlayer.h | 58 kWhatSetSource = 'ssrc',
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| /external/chromium_org/content/browser/resources/media/ |
| stats_table.js | 8 * @param {SsrcInfoManager} ssrcInfoManager The source of the ssrc info.
14 * @param {SsrcInfoManager} ssrcInfoManager The source of the ssrc info.
97 if (report.type == 'ssrc') {
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| webrtc_internals.css | 17 .ssrc-info-block {
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| /external/chromium_org/media/cast/audio_receiver/ |
| audio_decoder_unittest.cc | 63 rtp_header.webrtc.header.ssrc = 0x12345678;
113 rtp_header.webrtc.header.ssrc = 0x12345678;
182 rtp_header.webrtc.header.ssrc = 0x12345678;
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| /external/chromium_org/media/cast/rtcp/ |
| rtcp_sender_unittest.cc | 124 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
259 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
293 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
335 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
385 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
473 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
546 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender.
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| /external/chromium_org/third_party/libjingle/source/talk/examples/call/ |
| callclient.cc | 265 // TODO: Use a random ssrc
781 params.ssrc = stream.first_ssrc();
867 if (data_streams && GetStreamBySsrc(*data_streams, params.ssrc, &stream)) {
869 "Received data from '%s' on stream '%s' (ssrc=%u): %s",
871 params.ssrc, text.c_str());
874 "Received data (ssrc=%u): %s",
875 params.ssrc, text.c_str());
1474 uint32 ssrc = stream.first_ssrc(); local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| webrtcsdp.cc | 121 static const char kAttributeSsrc[] = "ssrc";
135 static const char kAttributeSsrcGroup[] = "ssrc-group";
546 // a=ssrc:<ssrc-id> <attribute>:<value>
1392 std::vector<uint32>::const_iterator ssrc = local
1401 uint32 ssrc = track->ssrcs[i]; local
2563 uint32 ssrc = ssrc_group->ssrcs.front(); local
2712 uint32 ssrc = talk_base::FromString<uint32>(fields[i]); local
[all...] |
| /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...] |
| /external/chromium/third_party/libjingle/source/talk/session/phone/ |
| mediasessionclient.h | 183 uint32 ssrc() const { return ssrc_; } function in class:cricket::MediaContentDescription
185 void set_ssrc(uint32 ssrc) {
186 ssrc_ = ssrc;
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| /external/chromium_org/chrome/renderer/media/ |
| cast_rtp_stream.cc | 69 config->sender_ssrc = payload_params.ssrc;
87 config->sender_ssrc = payload_params.ssrc;
215 ssrc(0),
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| cast_rtp_stream.h | 35 int ssrc; member in struct:CastRtpPayloadParams
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| /external/chromium_org/third_party/WebKit/Source/testing/runner/ |
| MockWebRTCPeerConnectionHandler.cpp | 277 size_t reportIndex = response.addReport("Mock video", "ssrc", currentDate);
281 size_t reportIndex = response.addReport("Mock audio", "ssrc", currentDate);
283 reportIndex = response.addReport("Mock video", "ssrc", currentDate);
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| /external/bluetooth/bluedroid/stack/avdt/ |
| avdt_scb_act.c | 67 ** Description This function generates a SSRC number unique to the stream.
69 ** Returns SSRC value.
243 UINT32 ssrc; local
255 BE_STREAM_TO_UINT32(ssrc, p);
330 UINT32 ssrc; local
344 BE_STREAM_TO_UINT32(ssrc, p);
373 AVDT_TRACE_WARNING5( " - SDES SSRC=0x%08x sc=%d %d len=%d %s",
374 ssrc, o_cc, *p, *(p+1), p+2);
414 UINT32 ssrc; local
567 BE_STREAM_TO_UINT32(ssrc, p_payload)
1204 UINT32 ssrc; local
1253 UINT32 ssrc; local
[all...] |
| /external/chromium_org/chrome/common/extensions/api/ |
| cast_streaming_rtp_stream.idl | 22 long? ssrc;
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| /external/chromium_org/media/cast/test/ |
| sender.cc | 100 test::InputBuilder input_tx("Choose audio sender SSRC.",
104 test::InputBuilder input_rx("Choose audio receiver SSRC.",
110 test::InputBuilder input_tx("Choose video sender SSRC.",
114 test::InputBuilder input_rx("Choose video receiver SSRC.",
|
