/external/chromium_org/media/cast/net/rtp/ |
H A D | rtp_sender.h | 67 uint32 ssrc() const { return config_.ssrc; } function in class:media::cast::RtpSender
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H A D | rtp_header_parser.cc | 29 ssrc(0), 65 uint32 rtp_timestamp, ssrc; local 67 big_endian_reader.ReadU32(&ssrc); 75 parsed_packet->ssrc = ssrc;
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H A D | rtp_header_parser.h | 32 uint32 ssrc; member in struct:media::cast::RtpCastTestHeader
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H A D | rtp_packetizer.h | 34 unsigned int ssrc; member in struct:media::cast::RtpPacketizerConfig
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | ssrc_database.cc | 58 uint32_t ssrc = GenerateRandom(); local 60 while(_ssrcMap.find(ssrc) != _ssrcMap.end()) 62 ssrc = GenerateRandom(); 64 _ssrcMap[ssrc] = 0; 66 return ssrc; 70 SSRCDatabase::RegisterSSRC(const uint32_t ssrc) argument 73 _ssrcMap[ssrc] = 0; 78 SSRCDatabase::ReturnSSRC(const uint32_t ssrc) argument 81 _ssrcMap.erase(ssrc); 108 uint32_t ssrc local [all...] |
H A D | forward_error_correction.h | 81 // The ssrc member is needed to ensure we can restore the SSRC field of 91 uint32_t ssrc; // SSRC of the current frame. Must be set for FEC member in class:webrtc::ForwardErrorCorrection::ReceivedPacket
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H A D | remote_ntp_time_estimator.cc | 29 bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(uint32_t ssrc, argument 33 rtp_rtcp->RTT(ssrc, &rtt, NULL, NULL, NULL);
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H A D | rtp_packet_history_unittest.cc | 42 void CreateRtpPacket(uint16_t seq_num, uint32_t ssrc, uint8_t payload, argument 52 array[(*cur_pos)++] = ssrc >> 24; 53 array[(*cur_pos)++] = ssrc >> 16; 54 array[(*cur_pos)++] = ssrc >> 8; 55 array[(*cur_pos)++] = ssrc;
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H A D | tmmbr_help.h | 42 return _data.at(i).ssrc; 65 SetElement() : tmmbr(0), packet_oh(0), ssrc(0) {} 68 uint32_t ssrc; member in class:webrtc::TMMBRSet::SetElement 94 bool IsOwner(const uint32_t ssrc, const uint32_t length) const;
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/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_remb_unittest.cc | 50 unsigned int ssrc = 1234; local 51 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 75 unsigned int ssrc = 1234; local 76 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 101 unsigned int ssrc[] = { 1234, 5678 }; local 102 std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeo 132 unsigned int ssrc[] = { 1234, 5678 }; local 166 unsigned int ssrc[] = { 1234, 5678 }; local 200 unsigned int ssrc = 1234; local 233 unsigned int ssrc = 1234; local [all...] |
H A D | encoder_state_feedback_unittest.cc | 37 void(uint32_t ssrc, uint8_t picture_id)); 39 void(uint32_t ssrc, uint64_t picture_id)); 57 const int ssrc = 1234; local 59 EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc, &encoder)); 61 EXPECT_CALL(encoder, OnReceivedIntraFrameRequest(ssrc)) 64 OnReceivedIntraFrameRequest(ssrc); 67 EXPECT_CALL(encoder, OnReceivedSLI(ssrc, sli_picture_id)) 70 ssrc, sli_picture_id); 73 EXPECT_CALL(encoder, OnReceivedRPSI(ssrc, rpsi_picture_id)) 76 ssrc, rpsi_picture_i 131 const int ssrc = 1234; local [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | rtpdump_unittest.cc | 50 uint32 ssrc; local 60 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc)); 61 EXPECT_EQ(kTestSsrc, ssrc); 131 uint32 ssrc; local 132 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc)); 133 EXPECT_EQ(kTestSsrc, ssrc); 138 // Rewind the stream and read again with a specified ssrc. 147 uint32 ssrc; local 148 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc)); 149 EXPECT_EQ(send_ssrc, ssrc); [all...] |
H A D | rtputils.h | 43 uint32 ssrc; member in struct:cricket::RtpHeader
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H A D | rtputils_unittest.cc | 97 uint32 ssrc; local 98 EXPECT_TRUE(GetRtpSsrc(kPcmuFrame, sizeof(kPcmuFrame), &ssrc)); 99 EXPECT_EQ(1u, ssrc); 106 EXPECT_EQ(1u, header.ssrc); 111 EXPECT_FALSE(GetRtpSsrc(kInvalidPacket, sizeof(kInvalidPacket), &ssrc)); 135 EXPECT_EQ(3333u, header.ssrc); 151 EXPECT_EQ(3333u, header.ssrc); 182 uint32 ssrc; local 185 &ssrc)); 188 &ssrc)); [all...] |
H A D | fakemediaprocessor.h | 48 virtual void OnFrame(uint32 ssrc, argument 53 virtual void OnFrame(uint32 ssrc, VideoFrame* frame_ptr, bool* drop_frame) { argument 60 virtual void OnVoiceMute(uint32 ssrc, bool muted) {} argument 61 virtual void OnVideoMute(uint32 ssrc, bool muted) {} argument
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | typingmonitor.cc | 63 void TypingMonitor::OnVoiceChannelError(uint32 ssrc, argument
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H A D | bundlefilter.cc | 45 // For rtcp packets, we check whether the ssrc can be found or is the special 61 // Rtcp packets using ssrc filter. 63 uint32 ssrc = 0; local 70 if (!GetRtcpSsrc(data, len, &ssrc)) return false; 71 if (ssrc == kSsrc01) { 79 return !HasStreams() || FindStream(ssrc); 95 bool BundleFilter::RemoveStream(uint32 ssrc) { argument 96 return RemoveStreamBySsrc(&streams_, ssrc); 103 bool BundleFilter::FindStream(uint32 ssrc) const { 104 if (ssrc [all...] |
H A D | currentspeakermonitor.cc | 94 uint32 ssrc = stream_list_it->first; local 95 active_ssrc_to_level_map[ssrc] = stream_list_it->second; 99 if (ssrc_to_speaking_state_map_.find(ssrc) == 101 ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
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/external/chromium_org/chrome/browser/media/ |
H A D | webrtc_browsertest_perf.cc | 14 // A ssrc stats key will be on the form stats.<bucket>-<key>.values. 24 const std::string& ssrc, const base::DictionaryValue& pc_dict) { 26 if (!pc_dict.GetString(Statistic("audioOutputLevel", ssrc), &value)) { 31 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesReceived", ssrc), &value)); 34 EXPECT_TRUE(pc_dict.GetString(Statistic("packetsLost", ssrc), &value)); 42 const std::string& ssrc, const base::DictionaryValue& pc_dict) { 44 if (!pc_dict.GetString(Statistic("audioInputLevel", ssrc), &value)) { 49 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesSent", ssrc), &value)); 52 EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value)); 55 EXPECT_TRUE(pc_dict.GetString(Statistic("googRtt", ssrc), 23 MaybePrintResultsForAudioReceive( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 41 MaybePrintResultsForAudioSend( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 61 MaybePrintResultsForVideoSend( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 119 MaybePrintResultsForVideoReceive( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 231 const std::string& ssrc = *ssrc_iterator; local [all...] |
/external/chromium_org/chrome/renderer/media/ |
H A D | cast_ipc_dispatcher.cc | 104 uint32 ssrc, 108 sender->OnRtt(ssrc, rtt); 116 uint32 ssrc, 120 sender->OnRtcpCastMessage(ssrc, cast_message); 103 OnRtt(int32 channel_id, uint32 ssrc, base::TimeDelta rtt) argument 114 OnRtcpCastMessage( int32 channel_id, uint32 ssrc, const media::cast::RtcpCastMessage& cast_message) argument
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/external/chromium_org/content/browser/resources/media/ |
H A D | 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/net/rtcp/ |
H A D | rtcp_defines.cc | 12 RtcpCastMessage::RtcpCastMessage(uint32 ssrc) argument 13 : media_ssrc(ssrc), ack_frame_id(0u), target_delay_ms(0) {}
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/external/chromium_org/third_party/libsrtp/srtp/test/ |
H A D | dtls_srtp_driver.c | 53 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc); 192 policy.ssrc.type = ssrc_any_inbound; 214 * srtp_create_test_packet(len, ssrc) returns a pointer to a 216 * by pkt_octet_len and the SSRC value ssrc. The total length of the 227 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc) { argument 247 hdr->ssrc = htonl(ssrc); /* synch. source */
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H A D | rtp.c | 120 octets_recvd, receiver->message.header.ssrc); 145 unsigned int ssrc) { 148 sender->message.header.ssrc = htonl(ssrc); 169 unsigned int ssrc) { 172 rcvr->message.header.ssrc = htonl(ssrc); 142 rtp_sender_init(rtp_sender_t sender, int sock, struct sockaddr_in addr, unsigned int ssrc) argument 166 rtp_receiver_init(rtp_receiver_t rcvr, int sock, struct sockaddr_in addr, unsigned int ssrc) argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api.cc | 114 unsigned int ssrc = 0; local 121 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type); 123 EXPECT_EQ(1u, ssrc); 129 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type); 133 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type); 146 rtx_header.ssrc = kRtxSsrc; 149 rtx_header.ssrc = 0; 151 rtx_header.ssrc = kRtxSsrc;
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