| /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 | 356 big_endian_reader.ReadU32(&field_.report_block_item.ssrc);
414 uint32 ssrc; local
417 big_endian_reader.ReadU32(&ssrc);
422 field_.c_name.sender_ssrc = ssrc;
895 big_endian_reader.ReadU32(&field_.fir_item.ssrc);
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| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| 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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| hybridvideoengine_unittest.cc | 373 EXPECT_EQ(0U, catcher.ssrc());
376 EXPECT_EQ(0U, catcher.ssrc());
382 EXPECT_EQ(1U, catcher.ssrc());
393 EXPECT_EQ(2U, catcher.ssrc());
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| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| channelmanager.h | 185 bool RegisterVoiceProcessor(uint32 ssrc,
188 bool UnregisterVoiceProcessor(uint32 ssrc,
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_receiver.h | 45 void SetRelaySSRC( const uint32_t ssrc);
46 int32_t SetRemoteSSRC( const uint32_t ssrc);
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| rtcp_sender.h | 97 void SetSSRC( const uint32_t ssrc);
99 void SetRemoteSSRC(uint32_t ssrc);
106 int32_t AddMixedCNAME(const uint32_t SSRC,
109 int32_t RemoveMixedCNAME(const uint32_t SSRC);
128 uint32_t SSRC,
131 int32_t RemoveExternalReportBlock(uint32_t SSRC);
142 const uint32_t* SSRC);
202 uint32_t SSRC,
296 uint32_t _remoteSSRC; // SSRC that we receive on our RTP channel
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| rtp_payload_registry.cc | 242 return rtx_ && ssrc_rtx_ == header.ssrc;
263 // Replace the SSRC and the sequence number with the originals.
285 void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) {
287 ssrc_rtx_ = ssrc;
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| rtp_receiver_impl.cc | 152 uint32_t RtpReceiverImpl::SSRC() const {
288 if (ssrc_ != rtp_header.ssrc ||
290 // We need the payload_type_ to make the call if the remote SSRC is 0.
299 // Do we have a SSRC? Then the stream is restarted.
319 ssrc_ = rtp_header.ssrc;
326 cb_rtp_feedback_->OnIncomingSSRCChanged(id_, rtp_header.ssrc);
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| fec_receiver_impl.cc | 173 received_packet->ssrc =
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| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/test/ |
| bwe_test_framework.cc | 542 float fps, uint32_t kbps, uint32_t ssrc,
556 prototype_header_.ssrc = ssrc;
599 uint32_t ssrc,
601 : VideoSender(flow_id, listener, fps, kbps, ssrc, first_frame_offset) {}
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| bwe_test_framework.h | 388 uint32_t kbps, uint32_t ssrc, float first_frame_offset);
416 float fps, uint32_t kbps, uint32_t ssrc,
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| /external/chromium_org/third_party/webrtc/voice_engine/ |
| channel.h | 320 int SetLocalSSRC(unsigned int ssrc);
321 int GetLocalSSRC(unsigned int& ssrc);
322 int GetRemoteSSRC(unsigned int& ssrc);
391 uint32_t ssrc);
396 void ResetStatistics(uint32_t ssrc);
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| /external/bluetooth/bluedroid/stack/avdt/ |
| avdt_scb_act.c | 68 ** Description This function generates a SSRC number unique to the stream.
70 ** Returns SSRC value.
250 UINT32 ssrc; local
262 BE_STREAM_TO_UINT32(ssrc, p);
337 UINT32 ssrc; local
351 BE_STREAM_TO_UINT32(ssrc, p);
380 AVDT_TRACE_WARNING( " - SDES SSRC=0x%08x sc=%d %d len=%d %s",
381 ssrc, o_cc, *p, *(p+1), p+2);
421 UINT32 ssrc; local
574 BE_STREAM_TO_UINT32(ssrc, p_payload)
1218 UINT32 ssrc; local
1265 UINT32 ssrc; local
[all...] |
| /external/chromium_org/third_party/webrtc/video/ |
| video_send_stream_tests.cc | 128 // set up and some frames are sent on a random-generated SSRC
129 // before the correct SSRC gets set.
130 // EXPECT_TRUE(valid_ssrcs_[header.ssrc])
131 // << "Received unknown SSRC: " << header.ssrc;
133 // if (!valid_ssrcs_[header.ssrc])
136 if (!is_observed_[header.ssrc]) {
137 is_observed_[header.ssrc] = true;
186 << (send_single_ssrc_first ? "first SSRC." : "SSRCs.");
367 virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE
1077 uint32_t ssrc = stats.substreams.begin()->first; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_impl_unittest.cc | 265 rtp_header.header.ssrc = kSsrc;
376 rtp_header.header.ssrc = 0x87654321;
418 rtp_header.header.ssrc = 0x87654321;
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_channel.cc | 218 // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
792 int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
797 rtp_rtcp_->SetRtxSsrc(SSRC);
799 rtp_rtcp_->SetSSRC(SSRC);
815 rtp_rtcp_module->SetRtxSsrc(SSRC);
817 rtp_rtcp_module->SetSSRC(SSRC);
823 const uint32_t SSRC) {
824 vie_receiver_.SetRtxSsrc(SSRC);
828 int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) {
869 uint32_t ssrc; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvoiceengine_unittest.cc | 110 void OnVoiceChannelError(uint32 ssrc,
112 ssrc_ = ssrc;
119 uint32 ssrc() const { function in class:WebRtcVoiceEngineTestFake::ChannelErrorListener
178 void TestInsertDtmf(uint32 ssrc, bool caller) {
196 EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111, cricket::DF_SEND));
203 EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123, cricket::DF_SEND));
208 // Check we fail if the ssrc is invalid.
213 EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123, cricket::DF_SEND));
218 EXPECT_TRUE(channel_->InsertDtmf(ssrc, 3, 134, cricket::DF_PLAY));
224 EXPECT_TRUE(channel_->InsertDtmf(ssrc, 4, 145
2226 unsigned int ssrc = 0; local
2469 unsigned int ssrc = 0; local
[all...] |
| /external/chromium_org/chrome/common/ |
| cast_messages.h | 49 IPC_STRUCT_TRAITS_MEMBER(ssrc)
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| /external/chromium_org/media/cast/video_sender/ |
| external_video_encoder_unittest.cc | 89 video_config_.rtp_config.ssrc = 1;
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| /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);
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| initial_delay_manager.cc | 29 last_packet_rtp_info_.header.ssrc = 0;
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| target_delay_unittest.cc | 40 rtp_info_.header.ssrc = 0x12345678;
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| /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.
90 'ssrc': true,
281 if (report.type == 'ssrc') {
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| /external/chromium_org/media/cast/receiver/ |
| frame_receiver.cc | 110 uint32* ssrc) {
113 return big_endian_reader.Skip(8) && big_endian_reader.ReadU32(ssrc);
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