/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 | 78 const uint32 ssrc = 7; local 79 cricket::StreamParams one_sp = cricket::StreamParams::CreateLegacy(ssrc); 81 EXPECT_EQ(ssrc, one_sp.first_ssrc()); 83 EXPECT_TRUE(one_sp.has_ssrc(ssrc)); 84 EXPECT_FALSE(one_sp.has_ssrc(ssrc+1));
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H A D | streamparams.cc | 145 bool GetStreamBySsrc(const StreamParamsVec& streams, uint32 ssrc, argument 147 return GetStream(streams, StreamSelector(ssrc), stream_out); 172 bool RemoveStreamBySsrc(StreamParamsVec* streams, uint32 ssrc) { argument 173 return RemoveStream(streams, StreamSelector(ssrc));
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H A D | streamparams.h | 71 static StreamParams CreateLegacy(uint32 ssrc) { argument 73 stream.ssrcs.push_back(ssrc); 101 bool has_ssrc(uint32 ssrc) const { 102 return std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end(); 104 void add_ssrc(uint32 ssrc) { argument 105 ssrcs.push_back(ssrc); 123 // Convenience function to add an FID ssrc for a primary_ssrc 129 // Convenience function to lookup the FID ssrc for a primary_ssrc. 159 // A Stream can be selected by either groupid+id or ssrc. 161 explicit StreamSelector(uint32 ssrc) argument 180 uint32 ssrc; member in struct:cricket::StreamSelector [all...] |
H A D | fakenetworkinterface.h | 77 int NumRtpBytes(uint32 ssrc) { argument 80 GetNumRtpBytesAndPackets(ssrc, &bytes, NULL); 89 int NumRtpPackets(uint32 ssrc) { argument 92 GetNumRtpBytesAndPackets(ssrc, NULL, &packets); 124 // Indicate that |n|'th packet for |ssrc| should be dropped. 125 void AddPacketDrop(uint32 ssrc, uint32 n) { argument 126 drop_map_[ssrc].insert(n); 206 void GetNumRtpBytesAndPackets(uint32 ssrc, int* bytes, int* packets) { argument 219 if (ssrc == cur_ssrc) { 235 // Map to track counts of packets that have been sent per ssrc [all...] |
H A D | filemediaengine.cc | 131 void SetSendSsrc(uint32 ssrc); 196 void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) { argument 198 rtp_dump_reader_->SetSsrc(ssrc); 232 uint32 ssrc; local 233 if (!packet->GetRtpSsrc(&ssrc)) { 238 first_ssrc_ = ssrc; 240 if (ssrc == first_ssrc_) { 286 bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) { argument 287 if (ssrc != send_ssrc_) 329 bool FileVideoChannel::RemoveSendStream(uint32 ssrc) { argument [all...] |
H A D | filemediaengine_unittest.cc | 187 uint32 ssrc; local 188 if (!packet.GetRtpSsrc(&ssrc)) { 191 ssrcs.insert(ssrc); 378 // Test that we can specify the ssrc for outgoing RTP packets.
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H A D | hybridvideoengine.cc | 108 bool HybridVideoMediaChannel::SetRenderer(uint32 ssrc, argument 112 ret = channel1_->SetRenderer(ssrc, renderer); 115 ret = channel2_->SetRenderer(ssrc, renderer); 131 bool HybridVideoMediaChannel::MuteStream(uint32 ssrc, bool muted) { argument 134 ret = channel1_->MuteStream(ssrc, muted); 137 ret = channel2_->MuteStream(ssrc, muted); 175 bool HybridVideoMediaChannel::SetSendStreamFormat(uint32 ssrc, argument 177 return active_channel_ && active_channel_->SetSendStreamFormat(ssrc, format); 219 bool HybridVideoMediaChannel::SetCapturer(uint32 ssrc, argument 223 ret = channel1_->SetCapturer(ssrc, capture 242 RemoveSendStream(uint32 ssrc) argument 258 RemoveRecvStream(uint32 ssrc) argument 346 OnMediaError(uint32 ssrc, Error error) argument [all...] |
H A D | rtpdataengine.cc | 180 << "' with ssrc=" << stream.first_ssrc() 186 // TODO(pthatcher): This should be per-stream, not per-ssrc. 193 << "' with ssrc=" << stream.first_ssrc(); 197 bool RtpDataMediaChannel::RemoveSendStream(uint32 ssrc) { argument 199 if (!GetStreamBySsrc(send_streams_, ssrc, &found_stream)) { 203 RemoveStreamBySsrc(&send_streams_, ssrc); 204 delete rtp_clock_by_send_ssrc_[ssrc]; 205 rtp_clock_by_send_ssrc_.erase(ssrc); 217 << "' with ssrc=" << stream.first_ssrc() 224 << "' with ssrc 228 RemoveRecvStream(uint32 ssrc) argument [all...] |
H A D | rtpdump.cc | 91 bool RtpDumpPacket::GetRtpSsrc(uint32* ssrc) const { 93 cricket::GetRtpSsrc(&data[0], data.size(), ssrc); 110 void RtpDumpReader::SetSsrc(uint32 ssrc) { argument 111 ssrc_override_ = ssrc; 151 // If the packet is RTP and we have specified a ssrc, replace the RTP ssrc 152 // with the specified ssrc.
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H A D | testutils.cc | 67 ret &= buf->ReadUInt32(&ssrc); 78 ssrc == ssc && 157 size_t count, talk_base::StreamInterface* stream, uint32 ssrc) { 193 ssrc); 156 VerifyTestPacketsFromStream( size_t count, talk_base::StreamInterface* stream, uint32 ssrc) argument
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/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/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, 209 if (report.type == 'ssrc') {
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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/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; 245 bool GetRemoteAudioTrackSsrc(const std::string& track_id, uint32* ssrc) const; 246 bool GetRemoteVideoTrackSsrc(const std::string& track_id, uint32* ssrc) const; 281 TrackInfo() : ssrc(0) {} 284 uint32 ssrc) 287 ssrc(ssrc) { 282 TrackInfo(const std::string& stream_label, const std::string track_id, uint32 ssrc) argument 291 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 | 96 uint32 ssrc) { 100 uint32 ssrc) { 104 uint32 ssrc) { 109 uint32 ssrc) { 94 OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream, webrtc::AudioTrackInterface* audio_track, uint32 ssrc) argument 98 OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream, webrtc::VideoTrackInterface* video_track, uint32 ssrc) argument 102 OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream, webrtc::AudioTrackInterface* audio_track, uint32 ssrc) argument 107 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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/external/chromium/third_party/libjingle/source/talk/session/phone/ |
H A D | filemediaengine.cc | 182 uint32 ssrc; local 183 if (!packet->GetRtpSsrc(&ssrc)) { 188 first_ssrc_ = ssrc; 190 if (ssrc == first_ssrc_) {
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