/external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/ |
H A D | rtp_packetizer_config.h | 26 unsigned int ssrc; member in struct:media::cast::RtpPacketizerConfig
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/external/chromium_org/media/cast/rtp_receiver/rtp_parser/ |
H A D | rtp_parser.h | 18 ssrc = 0; 23 uint32 ssrc; member in struct:media::cast::RtpParserConfig
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H A D | rtp_parser.cc | 33 rtp_header->webrtc.header.ssrc == parser_config_.ssrc) { 37 // Not a valid payload type / ssrc combination. 52 uint32 rtp_timestamp, ssrc; local 56 big_endian_reader.ReadU32(&ssrc); 58 if (ssrc != parser_config_.ssrc) return false; 64 rtp_header->webrtc.header.ssrc = ssrc;
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/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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H A D | streamparams_unittest.cc | 80 const uint32 ssrc = 7; local 81 cricket::StreamParams one_sp = cricket::StreamParams::CreateLegacy(ssrc); 83 EXPECT_EQ(ssrc, one_sp.first_ssrc()); 85 EXPECT_TRUE(one_sp.has_ssrc(ssrc)); 86 EXPECT_FALSE(one_sp.has_ssrc(ssrc+1)); 225 // stream1 has extra non-sim, non-fid ssrc.
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H A D | streamparams.h | 39 // StreamParams would then contain ssrc = {10,11,20,21,30,31} and 80 static StreamParams CreateLegacy(uint32 ssrc) { argument 82 stream.ssrcs.push_back(ssrc); 110 bool has_ssrc(uint32 ssrc) const { 111 return std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end(); 113 void add_ssrc(uint32 ssrc) { argument 114 ssrcs.push_back(ssrc); 132 // Convenience function to add an FID ssrc for a primary_ssrc 138 // Convenience function to lookup the FID ssrc for a primary_ssrc. 168 // A Stream can be selected by either groupid+id or ssrc 170 StreamSelector(uint32 ssrc) argument 189 uint32 ssrc; member in struct:cricket::StreamSelector [all...] |
/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | ssrcmuxfilter.cc | 50 uint32 ssrc = 0; local 52 GetRtpSsrc(data, len, &ssrc); 61 if (!GetRtcpSsrc(data, len, &ssrc)) return false; 62 if (ssrc == kSsrc01) { 70 return FindStream(ssrc); 82 bool SsrcMuxFilter::RemoveStream(uint32 ssrc) { argument 83 return RemoveStreamBySsrc(&streams_, ssrc); 86 bool SsrcMuxFilter::FindStream(uint32 ssrc) const { 87 if (ssrc == 0) { 90 return (GetStreamBySsrc(streams_, ssrc, NUL [all...] |
H A D | typingmonitor.cc | 63 void TypingMonitor::OnVoiceChannelError(uint32 ssrc, argument
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H A D | currentspeakermonitor.cc | 88 uint32 ssrc = stream_list_it->first; local 89 active_ssrc_to_level_map[ssrc] = stream_list_it->second; 93 if (ssrc_to_speaking_state_map_.find(ssrc) == 95 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)); 33 EXPECT_TRUE(pc_dict.GetString(Statistic("packetsLost", ssrc), &value)); 40 const std::string& ssrc, const base::DictionaryValue& pc_dict) { 42 if (!pc_dict.GetString(Statistic("audioInputLevel", ssrc), &value)) { 47 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesSent", ssrc), &value)); 49 EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value)); 52 EXPECT_TRUE(pc_dict.GetString(Statistic("googRtt", ssrc), 23 MaybePrintResultsForAudioReceive( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 39 MaybePrintResultsForAudioSend( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 57 MaybePrintResultsForVideoReceive( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 87 MaybePrintResultsForVideoSend( const std::string& ssrc, const base::DictionaryValue& pc_dict) argument 193 const std::string& ssrc = *ssrc_iterator; local [all...] |
/external/chromium_org/chrome/renderer/media/ |
H A D | cast_rtp_stream.h | 35 int ssrc; member in struct:CastRtpPayloadParams
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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. 85 'ssrc': true, 210 if (report.type == 'ssrc') {
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/external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/test/ |
H A D | rtp_header_parser.cc | 29 ssrc(0), 66 uint32 rtp_timestamp, ssrc; local 68 big_endian_reader.ReadU32(&ssrc); 76 parsed_packet->ssrc = ssrc;
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H A D | rtp_header_parser.h | 31 uint32 ssrc; member in struct:media::cast::RtpCastTestHeader
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/external/srtp/test/ |
H A D | dtls_srtp_driver.c | 53 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc); 185 policy.ssrc.type = ssrc_any_inbound; 201 * srtp_create_test_packet(len, ssrc) returns a pointer to a 203 * by pkt_octet_len and the SSRC value ssrc. The total length of the 214 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc) { argument 234 hdr->ssrc = htonl(ssrc); /* synch. source */
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H A D | rtp.c | 80 octets_recvd, receiver->message.header.ssrc); 105 unsigned int ssrc) { 108 sender->message.header.ssrc = htonl(ssrc); 129 unsigned int ssrc) { 132 rcvr->message.header.ssrc = htonl(ssrc); 102 rtp_sender_init(rtp_sender_t sender, int socket, struct sockaddr_in addr, unsigned int ssrc) argument 126 rtp_receiver_init(rtp_receiver_t rcvr, int socket, struct sockaddr_in addr, unsigned int ssrc) argument
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/external/chromium_org/media/cast/framer/ |
H A D | framer.cc | 16 uint32 ssrc, 21 &frame_id_map_, ssrc, decoder_faster_than_max_frame_rate, 14 Framer(base::TickClock* clock, RtpPayloadFeedback* incoming_payload_feedback, uint32 ssrc, bool decoder_faster_than_max_frame_rate, int max_unacked_frames) argument
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/external/chromium_org/media/cast/rtp_receiver/rtp_parser/test/ |
H A D | rtp_packet_builder.cc | 67 void RtpPacketBuilder::SetSsrc(uint32 ssrc) { argument 68 ssrc_ = ssrc;
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
H A D | mediastreamhandler.h | 50 TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc); 57 uint32 ssrc() const { return ssrc_; } function in class:webrtc::TrackHandler 76 uint32 ssrc, 97 uint32 ssrc, 117 uint32 ssrc, 137 uint32 ssrc, 160 virtual void AddAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc) = 0; 161 virtual void AddVideoTrack(VideoTrackInterface* video_track, uint32 ssrc) = 0; 183 uint32 ssrc) OVERRIDE; 185 uint32 ssrc) OVERRID [all...] |
H A D | mediastreamsignaling.h | 68 uint32 ssrc) = 0; 73 uint32 ssrc) = 0; 86 uint32 ssrc) = 0; 91 uint32 ssrc) = 0; 242 bool GetRemoteAudioTrackSsrc(const std::string& track_id, uint32* ssrc) const; 243 bool GetRemoteVideoTrackSsrc(const std::string& track_id, uint32* ssrc) const; 278 TrackInfo() : ssrc(0) {} 281 uint32 ssrc) 284 ssrc(ssrc) { 279 TrackInfo(const std::string& stream_label, const std::string track_id, uint32 ssrc) argument 288 uint32 ssrc; member in struct:webrtc::MediaStreamSignaling::TrackInfo [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
H A D | fakemediastreamsignaling.h | 97 uint32 ssrc) { 101 uint32 ssrc) { 105 uint32 ssrc) { 110 uint32 ssrc) { 95 OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream, webrtc::AudioTrackInterface* audio_track, uint32 ssrc) argument 99 OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream, webrtc::VideoTrackInterface* video_track, uint32 ssrc) argument 103 OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream, webrtc::AudioTrackInterface* audio_track, uint32 ssrc) argument 108 OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream, webrtc::VideoTrackInterface* video_track, uint32 ssrc) argument
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/external/chromium/third_party/libjingle/source/talk/p2p/base/ |
H A D | sessionmessages.h | 186 uint32 ssrc; member in struct:cricket::VideoViewRequest 191 VideoViewRequest(const std::string& nick_name, uint32 ssrc, uint32 width, argument 193 nick_name(nick_name), ssrc(ssrc), width(width), height(height),
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