| /external/chromium_org/third_party/webrtc/test/ |
| rtcp_packet_parser.h | 44 uint32_t Ssrc() const { return sr_.SenderSSRC; }
67 uint32_t Ssrc() const { return rr_.SenderSSRC; }
85 uint32_t Ssrc() const { return rb_.SSRC; }
149 uint32_t Ssrc() const { return cname_.SenderSSRC; }
168 uint32_t Ssrc() const { return bye_.SenderSSRC; }
186 uint32_t Ssrc() const { return rpsi_.SenderSSRC; }
246 uint32_t Ssrc() const { return pli_.SenderSSRC; }
265 uint32_t Ssrc() const { return sli_.SenderSSRC; }
304 uint32_t Ssrc() const { return fir_.SenderSSRC;
[all...] |
| /external/chromium_org/third_party/webrtc/video/ |
| call.cc | 252 receive_ssrcs_[it->second.ssrc] = receive_stream;
265 // separate SSRC there can be either one or two.
330 receive_ssrcs_.find(header.ssrc);
|
| video_receive_stream.cc | 73 assert(it->second.ssrc != 0);
76 rtp_rtcp_->SetRemoteSSRCType(channel_, kViEStreamTypeRtx, it->second.ssrc);
|
| rampup_tests.cc | 140 if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) {
150 rtx_media_ssrcs_[header.ssrc],
156 rtp_rtcp_->SetRemoteSSRC(header.ssrc);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_impl.cc | 123 ", ssrc=" << rtp_header.header.ssrc <<
142 ", ssrc=" << rtp_header.header.ssrc;
426 rtp_header.header.ssrc != ssrc_) {
429 LOG_F(LS_ERROR) << "Changing codec, SSRC or first packet "
446 packet->header.ssrc = rtp_header.header.ssrc;
465 // Reinitialize NetEq if it's needed (changed SSRC or first call).
466 if ((main_header.ssrc != ssrc_) || first_packet_)
[all...] |
| neteq_unittest.cc | 409 rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
421 rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
582 rtp_info.header.ssrc = 0x1234; // Just an arbitrary SSRC.
616 // only for 100 frames. Note the new SSRC, causing NetEQ to reset.
623 rtp_info.header.ssrc = 0x1235; // Just an arbitrary SSRC.
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| fakewebrtcvideoengine.h | 298 ssrcs_[0] = 0; // default ssrc.
399 // ssrcs_[0] is the default local ssrc.
[all...] |
| fakewebrtcvoiceengine.h | 769 WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
771 channels_[channel]->send_ssrc = ssrc;
774 WEBRTC_FUNC(GetLocalSSRC, (int channel, unsigned int& ssrc)) {
776 ssrc = channels_[channel]->send_ssrc;
779 WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
[all...] |
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| channel.cc | 57 StatisticsProxy(uint32_t ssrc)
59 ssrc_(ssrc) {}
63 uint32_t ssrc) OVERRIDE {
64 if (ssrc != ssrc_)
299 Channel::OnIncomingSSRCChanged(int32_t id, uint32_t ssrc)
302 "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
303 id, ssrc);
305 // Update ssrc so that NTP for AV sync can be updated.
306 _rtpRtcpModule->SetRemoteSSRC(ssrc);
318 void Channel::ResetStatistics(uint32_t ssrc) {
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| 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_org/third_party/libjingle/source/talk/session/media/ |
| 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));
772 buzz::XmlElement* CreateGingleSsrcElem(const buzz::QName& name, uint32 ssrc) {
774 if (ssrc) {
775 SetXmlBody(elem, ssrc);
[all...] |
| channelmanager.cc | 787 // is done, we will access the capturer using the ssrc (similar to how the
814 uint32 ssrc,
819 ssrc, processor, direction));
823 uint32 ssrc,
828 media_engine_.get(), ssrc, processor, direction));
|
| /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/chrome/renderer/extensions/ |
| cast_streaming_native_handler.cc | 74 cast_params->ssrc = ext_params.ssrc;
110 ext_params->ssrc = cast_params.ssrc;
|
| /external/chromium_org/media/cast/test/ |
| cast_benchmarks.cc | 285 audio_sender_config_.rtp_config.ssrc = 1;
297 audio_receiver_config_.incoming_ssrc = audio_sender_config_.rtp_config.ssrc;
306 video_sender_config_.rtp_config.ssrc = 3;
331 video_receiver_config_.incoming_ssrc = video_sender_config_.rtp_config.ssrc;
|
| end2end_unittest.cc | 465 audio_sender_config_.rtp_config.ssrc = 1;
477 audio_receiver_config_.incoming_ssrc = audio_sender_config_.rtp_config.ssrc;
489 video_sender_config_.rtp_config.ssrc = 3;
508 video_receiver_config_.incoming_ssrc = video_sender_config_.rtp_config.ssrc;
[all...] |
| /external/chromium_org/media/cast/video_sender/ |
| video_encoder_impl_unittest.cc | 69 video_config_.rtp_config.ssrc = 1;
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| forward_error_correction.cc | 70 uint32_t ssrc; // SSRC of the current frame. member in class:webrtc::FecPacket
537 fec_packet->ssrc = rx_packet->ssrc;
643 // Set the SSRC field.
645 fec_packet->ssrc);
|
| rtp_receiver_video.cc | 135 rtp_header->header.ssrc);
|
| rtp_utility.cc | 319 uint32_t SSRC = *ptr++ << 24;
320 SSRC += *ptr++ << 16;
321 SSRC += *ptr++ << 8;
322 SSRC += *ptr++;
325 header->ssrc = SSRC;
359 uint32_t SSRC = *ptr++ << 24;
360 SSRC += *ptr++ << 16;
361 SSRC += *ptr++ << 8;
362 SSRC += *ptr++
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| test_callbacks.cc | 92 rtpInfo.header.ssrc = 0;
|
| vcm_payload_sink_factory.cc | 173 new FileOutputFrameReceiver(base_out_filename_, stream->ssrc()));
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| webrtcsdp.cc | 123 static const char kAttributeSsrc[] = "ssrc";
134 static const char kAttributeSsrcGroup[] = "ssrc-group";
544 // a=ssrc:<ssrc-id> <attribute>:<value>
1430 std::vector<uint32>::const_iterator ssrc = local
1439 uint32 ssrc = track->ssrcs[i]; local
2608 uint32 ssrc = ssrc_group->ssrcs.front(); local
2760 uint32 ssrc = 0; local
[all...] |
| /external/chromium_org/media/cast/rtcp/ |
| rtcp_unittest.cc | 86 virtual bool SendRtcpPacket(uint32 ssrc,
178 config.rtp.config.ssrc = kSenderSsrc;
|
| /external/chromium_org/media/cast/audio_sender/ |
| audio_sender.cc | 55 audio_config.rtp_config.ssrc,
|
