/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
streamparams.h | 28 // This file contains structures for describing SSRCs from a media source such 32 // SsrcGroup is used to describe the relationship between the SSRCs that 60 SsrcGroup(const std::string& usage, const std::vector<uint32>& ssrcs) 61 : semantics(usage), ssrcs(ssrcs) { 65 return (semantics == other.semantics && ssrcs == other.ssrcs); 76 std::vector<uint32> ssrcs; // SSRCs of this type. member in struct:cricket::SsrcGroup 82 stream.ssrcs.push_back(ssrc) 162 std::vector<uint32> ssrcs; \/\/ All SSRCs for this source member in struct:cricket::StreamParams [all...] |
filemediaengine_unittest.cc | 188 std::set<uint32> ssrcs; local 195 ssrcs.insert(ssrc); 198 *ssrc_count = ssrcs.size(); 418 // Test the sender thread of the channel, where the input rtpdump has two SSRCs. 424 // with different SSRCs. 438 // these packets have the same SSRCs (that is, the packets with different 439 // SSRCs are skipped by the filemediaengine).
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videoengine_unittest.h | 531 // SetUp() already added kSsrc make sure duplicate SSRCs cant be added. 888 EXPECT_EQ(1U, info.senders[0].ssrcs().size()); 889 EXPECT_EQ(1U, info.receivers[0].ssrcs().size()); 890 EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]); 923 std::vector<uint32> ssrcs; local 1290 std::vector<uint32> ssrcs; local [all...] |
mediachannel.h | 582 // Creates a new outgoing media stream with SSRCs and CNAME as described 587 // multiple SSRCs. 589 // Creates a new incoming media stream with SSRCs and CNAME as described 594 // multiple SSRCs. 674 // media. (SSRCs shared between media streams can't be represented.) 715 std::vector<uint32> ssrcs() const { function in struct:cricket::MediaSenderInfo 774 std::vector<uint32> ssrcs() const { function in struct:cricket::MediaReceiverInfo [all...] |
/external/chromium_org/third_party/webrtc/video/ |
rampup_tests.cc | 28 std::vector<uint32_t> ssrcs; local 30 ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i)); 31 return ssrcs; 92 const std::vector<unsigned int>& ssrcs, unsigned int bitrate) { 112 bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); 233 const std::vector<unsigned int>& ssrcs, 237 bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); 372 std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100)) [all...] |
video_send_stream.cc | 45 ss << "{ssrcs: {"; 46 for (size_t i = 0; i < ssrcs.size(); ++i) { 47 ss << ssrcs[i]; local 48 if (i != ssrcs.size() - 1) 60 ss << "{ssrcs: {"; 61 for (size_t i = 0; i < ssrcs.size(); ++i) { 62 ss << ssrcs[i]; local 63 if (i != ssrcs.size() - 1) 84 if (rtx.payload_type != 0 || !rtx.ssrcs.empty()) 137 assert(config_.rtp.ssrcs.size() > 0) [all...] |
end_to_end_tests.cc | 493 // retransmitted and renders. Retransmission SSRCs are also checked. 549 send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); 923 // Test sets up a Call multiple senders with different resolutions and SSRCs. 1478 const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs; local [all...] |
/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/automated/ |
vie_network_test.cc | 82 uint8_t ssrcs = buf[4]; local 88 if (length < (8 + 4 * ssrcs)) { 92 for (uint8_t i = 0; i < ssrcs; ++i) {
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/external/chromium_org/third_party/webrtc/ |
video_send_stream.h | 80 std::vector<uint32_t> ssrcs; member in struct:webrtc::VideoSendStream::Config::Rtp 104 // SSRCs to use for the RTX streams. 105 std::vector<uint32_t> ssrcs; member in struct:webrtc::VideoSendStream::Config::Rtp::Rtx 153 // Set which streams to send. Must have at least as many SSRCs as configured
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
mediamessages.cc | 224 std::vector<uint32>* ssrcs, 235 ssrcs->push_back(ssrc); 248 std::vector<uint32> ssrcs; local 249 if (!ParseSsrcs(group_elem, &ssrcs, error)) { 252 ssrc_groups->push_back(SsrcGroup(semantics, ssrcs)); 267 if (!ParseSsrcs(stream_elem, &(stream.ssrcs), error)) { 320 void WriteSsrcs(const std::vector<uint32>& ssrcs, 322 for (std::vector<uint32>::const_iterator ssrc = ssrcs.begin(); 323 ssrc != ssrcs.end(); ++ssrc) { 339 WriteSsrcs(group->ssrcs, group_elem) [all...] |
mediasession.cc | 249 // The generated values are added to |ssrcs|. 253 std::vector<uint32>* ssrcs) { 259 std::count(ssrcs->begin(), ssrcs->end(), candidate) > 0); 260 ssrcs->push_back(candidate); 444 std::vector<uint32> ssrcs; local 446 GenerateSctpSids(*current_streams, &ssrcs); 449 GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs); 452 content_description->AddLegacyStream(ssrcs[0], ssrcs[1]) 479 std::vector<uint32> ssrcs; local [all...] |
/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
remote_bitrate_estimator_single_stream.cc | 53 // estimated payload bitrate in bits per second. |ssrcs| is the list of ssrcs 55 virtual bool LatestEstimate(std::vector<unsigned int>* ssrcs, 84 void GetSsrcs(std::vector<unsigned int>* ssrcs) const; 199 std::vector<unsigned int> ssrcs; local 200 GetSsrcs(&ssrcs); 201 observer_->OnReceiveBitrateChanged(ssrcs, target_bitrate); 220 std::vector<unsigned int>* ssrcs, 227 GetSsrcs(ssrcs); 228 if (ssrcs->empty() [all...] |
remote_bitrate_estimator_unittest_helper.cc | 22 const std::vector<unsigned int>& ssrcs, 313 std::vector<unsigned int> ssrcs; local 314 EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); 315 EXPECT_EQ(0u, ssrcs.size()); 318 EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); 326 EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); 327 EXPECT_EQ(0u, ssrcs.size()); 340 EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); 341 ASSERT_EQ(1u, ssrcs.size()); 342 EXPECT_EQ(kDefaultSsrc, ssrcs.front()) 500 std::vector<unsigned int> ssrcs; local [all...] |
/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/test/ |
bwe_test.cc | 121 virtual void OnReceiveBitrateChanged(const vector<unsigned int>& ssrcs, 128 vector<unsigned int> ssrcs; local 130 if (!estimator_->LatestEstimate(&ssrcs, &bps)) {
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtcp_packet_unittest.cc | 616 std::vector<uint32_t> ssrcs = parser.remb_item()->last_ssrc_list(); local 617 EXPECT_EQ(kRemoteSsrc, ssrcs[0]); 618 EXPECT_EQ(kRemoteSsrc + 1, ssrcs[1]); 619 EXPECT_EQ(kRemoteSsrc + 2, ssrcs[2]);
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rtcp_receiver_unittest.cc | 189 std::set<uint32_t> ssrcs; local 190 ssrcs.insert(kSourceSsrc); 191 rtcp_receiver_->SetSsrcs(kSourceSsrc, ssrcs); 208 std::set<uint32_t> ssrcs; local 209 ssrcs.insert(kSourceSsrc); 210 rtcp_receiver_->SetSsrcs(kSourceSsrc, ssrcs); 230 std::set<uint32_t> ssrcs(kSourceSsrcs, kSourceSsrcs + kNumSsrcs); 231 rtcp_receiver_->SetSsrcs(kSourceSsrcs[0], ssrcs); 362 std::set<uint32_t> ssrcs; local 363 ssrcs.insert(kSourceSsrc) 375 std::set<uint32_t> ssrcs; local 388 std::set<uint32_t> ssrcs; local 401 std::set<uint32_t> ssrcs; local 431 std::set<uint32_t> ssrcs; local 451 std::set<uint32_t> ssrcs; local 481 std::set<uint32_t> ssrcs; local 498 std::set<uint32_t> ssrcs; local 516 std::set<uint32_t> ssrcs; local 536 std::set<uint32_t> ssrcs; local 562 std::set<uint32_t> ssrcs; local 641 std::set<uint32_t> ssrcs; local 711 std::set<uint32_t> ssrcs; local 748 std::set<uint32_t> ssrcs; local 758 std::set<uint32_t> ssrcs; local 779 std::set<uint32_t> ssrcs; local 848 std::set<uint32_t> ssrcs; local [all...] |
rtp_rtcp_impl.cc | 206 std::vector<unsigned int> ssrcs; local 207 if (remote_bitrate_->LatestEstimate(&ssrcs, &target_bitrate)) { 208 if (!ssrcs.empty()) { 209 target_bitrate = target_bitrate / ssrcs.size(); 1330 std::set<uint32_t> ssrcs; local [all...] |
/external/chromium_org/third_party/webrtc/video_engine/ |
vie_channel_manager.cc | 123 std::list<unsigned int> ssrcs; local 124 ssrcs.push_back(ssrc); 125 vie_encoder->SetSsrcs(ssrcs); 368 const std::list<unsigned int>& ssrcs) { 382 for (std::list<unsigned int>::const_iterator it = ssrcs.begin(); 383 it != ssrcs.end(); ++it) { 417 std::vector<unsigned int> ssrcs; local 419 &ssrcs, estimated_bandwidth) || ssrcs.empty()) {
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vie_codec_impl.cc | 227 // Update all SSRCs to ViEEncoder. 228 std::list<unsigned int> ssrcs; local 234 ssrcs.push_back(ssrc); 242 ssrcs.push_back(ssrc); 245 vie_encoder->SetSsrcs(ssrcs); 246 shared_data_->channel_manager()->UpdateSsrcs(video_channel, ssrcs);
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
webrtcvideoengine2_unittest.cc | 820 const std::vector<uint32> ssrcs = MAKE_VECTOR(kSsrcs1); local 823 cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); 825 ASSERT_EQ(rtx_ssrcs.size(), send_stream->GetConfig().rtp.rtx.ssrcs.size()); 827 EXPECT_EQ(rtx_ssrcs[i], send_stream->GetConfig().rtp.rtx.ssrcs[i]); 831 cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); 833 << "No SSRCs for RTX configured by AddRecvStream."; 1143 const std::vector<uint32> ssrcs = MAKE_VECTOR(kSsrcs1); local 1155 const std::vector<uint32> ssrcs = MAKE_VECTOR(kSsrcs1); local 1229 const std::vector<uint32> ssrcs = MAKE_VECTOR(kSsrcs1); local [all...] |
webrtcvideoengine.cc | 4180 std::vector<uint32> ssrcs = sim_group->ssrcs; local [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
webrtcsdp.cc | 2811 std::vector<uint32> ssrcs; local [all...] |