/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...] |
/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...] |
H A D | cast_transport_host_filter.h | 37 void SendRtt(int32 channel_id, uint32 ssrc, base::TimeDelta rtt); 39 uint32 ssrc, 54 uint32 ssrc, 58 uint32 ssrc, 61 void OnCancelSendingFrames(int32 channel_id, uint32 ssrc, 63 void OnResendFrameForKickstart(int32 channel_id, uint32 ssrc,
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/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | encoder_state_feedback.cc | 30 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) { argument 31 owner_->OnReceivedIntraFrameRequest(ssrc); 33 virtual void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) { argument 34 owner_->OnReceivedSLI(ssrc, picture_id); 36 virtual void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) { argument 37 owner_->OnReceivedRPSI(ssrc, picture_id); 56 bool EncoderStateFeedback::AddEncoder(uint32_t ssrc, ViEEncoder* encoder) { argument 58 if (encoders_.find(ssrc) != encoders_.end()) { 59 // Two encoders must not have the same ssrc. 63 encoders_[ssrc] 83 OnReceivedIntraFrameRequest(uint32_t ssrc) argument 92 OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) argument 101 OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) argument [all...] |
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.h | 37 // Adds an encoder to receive feedback for a unique ssrc. 38 bool AddEncoder(uint32_t ssrc, ViEEncoder* encoder); 49 void OnReceivedIntraFrameRequest(uint32_t ssrc); 50 void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id); 51 void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id); 62 // Maps a unique ssrc to the given encoder.
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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/app/webrtc/ |
H A D | mediastreamprovider.h | 47 // Enable/disable the audio playout of a remote audio track with |ssrc|. 48 virtual void SetAudioPlayout(uint32 ssrc, bool enable, 50 // Enable/disable sending audio on the local audio track with |ssrc|. 52 virtual void SetAudioSend(uint32 ssrc, bool enable, 56 // Sets the audio playout volume of a remote audio track with |ssrc|. 58 virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) = 0; 69 virtual bool SetCaptureDevice(uint32 ssrc, 71 // Enable/disable the video playout of a remote video track with |ssrc|. 72 virtual void SetVideoPlayout(uint32 ssrc, bool enable, 74 // Enable sending video on the local video track with |ssrc| [all...] |
H A D | mediastreamhandler.cc | 36 TrackHandler::TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc) argument 38 ssrc_(ssrc), 87 uint32 ssrc, 89 : TrackHandler(track, ssrc), 107 provider_->SetAudioSend(ssrc(), false, options, NULL); 124 provider_->SetAudioSend(ssrc(), audio_track_->enabled(), options, renderer); 129 uint32 ssrc, 131 : TrackHandler(track, ssrc), 143 provider_->SetAudioPlayout(ssrc(), false, NULL); 150 provider_->SetAudioPlayout(ssrc(), audio_track 85 LocalAudioTrackHandler( AudioTrackInterface* track, uint32 ssrc, AudioProviderInterface* provider) argument 127 RemoteAudioTrackHandler( AudioTrackInterface* track, uint32 ssrc, AudioProviderInterface* provider) argument 162 LocalVideoTrackHandler( VideoTrackInterface* track, uint32 ssrc, VideoProviderInterface* provider) argument 195 RemoteVideoTrackHandler( VideoTrackInterface* track, uint32 ssrc, VideoProviderInterface* provider) argument 286 AddAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc) argument 295 AddVideoTrack(VideoTrackInterface* video_track, uint32 ssrc) argument 314 AddAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc) argument 322 AddVideoTrack(VideoTrackInterface* video_track, uint32 ssrc) argument 362 AddRemoteAudioTrack( MediaStreamInterface* stream, AudioTrackInterface* audio_track, uint32 ssrc) argument 374 AddRemoteVideoTrack( MediaStreamInterface* stream, VideoTrackInterface* video_track, uint32 ssrc) argument 404 AddLocalAudioTrack( MediaStreamInterface* stream, AudioTrackInterface* audio_track, uint32 ssrc) argument 416 AddLocalVideoTrack( MediaStreamInterface* stream, VideoTrackInterface* video_track, uint32 ssrc) argument [all...] |
/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
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 | bundlefilter.h | 47 // this is decided based on the sender ssrc values. 63 bool RemoveStream(uint32 ssrc); 68 bool FindStream(uint32 ssrc) const;
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/external/chromium_org/third_party/webrtc/video/ |
H A D | send_statistics_proxy.cc | 50 StreamStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) { argument 51 std::map<uint32_t, StreamStats>::iterator it = stats_.substreams.find(ssrc); 55 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) == 59 ssrc) == config_.rtp.rtx.ssrcs.end()) { 63 return &stats_.substreams[ssrc]; // Insert new entry and return ptr. 67 uint32_t ssrc) { 69 StreamStats* stats = GetStatsEntry(ssrc); 78 uint32_t ssrc) { 80 StreamStats* stats = GetStatsEntry(ssrc); 88 uint32_t ssrc) { 66 StatisticsUpdated(const RtcpStatistics& statistics, uint32_t ssrc) argument 76 DataCountersUpdated( const StreamDataCounters& counters, uint32_t ssrc) argument 87 Notify(const BitrateStatistics& bitrate, uint32_t ssrc) argument 97 FrameCountUpdated(FrameType frame_type, uint32_t frame_count, const unsigned int ssrc) argument 119 SendSideDelayUpdated(int avg_delay_ms, int max_delay_ms, uint32_t ssrc) argument [all...] |
H A D | send_statistics_proxy_unittest.cc | 95 const uint32_t ssrc = *it; local 96 StreamStats& ssrc_stats = expected_.substreams[ssrc]; 99 uint32_t offset = ssrc * sizeof(RtcpStatistics); 104 callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc); 109 const uint32_t ssrc = *it; local 110 StreamStats& ssrc_stats = expected_.substreams[ssrc]; 113 uint32_t offset = ssrc * sizeof(RtcpStatistics); 118 callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc); 157 const uint32_t ssrc = *it; local 159 StreamStats& stats = expected_.substreams[ssrc]; 169 const uint32_t ssrc = *it; local 188 const uint32_t ssrc = *it; local 203 const uint32_t ssrc = *it; local 225 const uint32_t ssrc = *it; local 235 const uint32_t ssrc = *it; local 252 const uint32_t ssrc = *it; local 264 const uint32_t ssrc = *it; local [all...] |
H A D | send_statistics_proxy.h | 43 uint32_t ssrc) OVERRIDE; 46 uint32_t ssrc) OVERRIDE; 49 virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) OVERRIDE; 54 const unsigned int ssrc) OVERRIDE; 75 uint32_t ssrc) OVERRIDE; 78 StreamStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
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/external/chromium_org/content/browser/resources/media/ |
H A D | ssrc_info_manager.js | 8 * Get the ssrc if |report| is an ssrc report. 14 * @return {?string} The ssrc. 17 if (report.type != 'ssrc') { 18 console.warn("Trying to get ssrc from non-ssrc report."); 22 // If the 'ssrc' name-value pair exists, return the value; otherwise, return 24 // The 'ssrc' name-value pair only exists in an upcoming Libjingle change. Old 25 // versions use id to refer to the ssrc. 31 if (report.stats.values[i] == 'ssrc') { [all...] |
/external/chromium_org/chrome/renderer/media/ |
H A D | cast_transport_sender_ipc.cc | 40 clients_[config.ssrc].cast_message_cb = cast_message_cb; 41 clients_[config.ssrc].rtt_cb = rtt_cb; 49 clients_[config.ssrc].cast_message_cb = cast_message_cb; 50 clients_[config.ssrc].rtt_cb = rtt_cb; 54 void CastTransportSenderIPC::InsertFrame(uint32 ssrc, argument 56 Send(new CastHostMsg_InsertFrame(channel_id_, ssrc, frame)); 60 uint32 ssrc, 64 ssrc, 70 uint32 ssrc, const std::vector<uint32>& frame_ids) { 72 ssrc, 59 SendSenderReport( uint32 ssrc, base::TimeTicks current_time, uint32 current_time_as_rtp_timestamp) argument 69 CancelSendingFrames( uint32 ssrc, const std::vector<uint32>& frame_ids) argument 76 ResendFrameForKickstart( uint32 ssrc, uint32 frame_id) argument 94 OnRtt(uint32 ssrc, base::TimeDelta rtt) argument 104 OnRtcpCastMessage( uint32 ssrc, const media::cast::RtcpCastMessage& cast_message) argument [all...] |
H A D | cast_transport_sender_ipc.h | 40 virtual void InsertFrame(uint32 ssrc, 43 uint32 ssrc, 47 uint32 ssrc, 49 virtual void ResendFrameForKickstart(uint32 ssrc, uint32 frame_id) OVERRIDE; 55 void OnRtt(uint32 ssrc, base::TimeDelta rtt); 56 void OnRtcpCastMessage(uint32 ssrc,
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/external/chromium_org/third_party/webrtc/modules/pacing/ |
H A D | paced_sender_unittest.cc | 29 bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, 39 bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, argument 68 uint32_t ssrc, uint16_t sequence_number, 71 EXPECT_FALSE(send_bucket_->SendPacket(priority, ssrc, 74 ssrc, sequence_number, capture_time_ms, false)) 85 uint32_t ssrc = 12345; local 88 SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, 90 SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, 92 SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++, 95 EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc, 67 SendAndExpectPacket(PacedSender::Priority priority, uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, int size, bool retransmission) argument 120 uint32_t ssrc = 12345; local 160 uint32_t ssrc = 12345; local 210 uint32_t ssrc = 12345; local 233 uint32_t ssrc = 12345; local 277 uint32_t ssrc = 12345; local 295 uint32_t ssrc = 12345; local 321 uint32_t ssrc = 12346; local 370 uint32_t ssrc = 12346; local 438 uint32_t ssrc = 12346; local 498 uint32_t ssrc = 12346; local 528 uint32_t ssrc = 12346; local [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
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 | videoprocessor.h | 46 virtual void OnFrame(uint32 ssrc, VideoFrame* frame, bool* drop_frame) = 0;
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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. 178 // A Stream can be selected by either groupid+id or ssrc 180 StreamSelector(uint32 ssrc) argument 199 uint32 ssrc; member in struct:cricket::StreamSelector [all...] |
H A D | fakemediaengine.h | 124 virtual bool RemoveSendStream(uint32 ssrc) { argument 125 return RemoveStreamBySsrc(&send_streams_, ssrc); 135 virtual bool RemoveRecvStream(uint32 ssrc) { argument 136 return RemoveStreamBySsrc(&receive_streams_, ssrc); 138 virtual bool MuteStream(uint32 ssrc, bool on) { argument 139 if (!HasSendStream(ssrc) && ssrc != 0) 142 muted_streams_.insert(ssrc); 144 muted_streams_.erase(ssrc); 147 bool IsStreamMuted(uint32 ssrc) cons 228 DtmfInfo(uint32 ssrc, int event_code, int duration, int flags) argument 231 uint32 ssrc; member in struct:cricket::FakeVoiceMediaChannel::DtmfInfo 292 RemoveRecvStream(uint32 ssrc) argument 298 SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) argument 318 SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) argument 348 PlayRingbackTone(uint32 ssrc, bool play, bool loop) argument 365 InsertDtmf(uint32 ssrc, int event_code, int duration, int flags) argument 371 SetOutputScaling(uint32 ssrc, double left, double right) argument 386 GetOutputScaling(uint32 ssrc, double* left, double* right) argument 395 GetLastMediaError(uint32* ssrc, VoiceMediaChannel::Error* error) argument 403 TriggerError(uint32 ssrc, VoiceMediaChannel::Error error) argument 467 CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info, uint32 ssrc, int event_code, int duration, int flags) argument 494 GetSendStreamFormat(uint32 ssrc, VideoFormat* format) argument 501 SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) argument 516 RemoveSendStream(uint32 ssrc) argument 553 SetRenderer(uint32 ssrc, VideoRenderer* r) argument 564 SetCapturer(uint32 ssrc, VideoCapturer* capturer) argument 585 RemoveRecvStream(uint32 ssrc) argument 618 SetSendStreamDefaultFormat(uint32 ssrc) argument 686 RemoveRecvStream(uint32 ssrc) argument 829 RegisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor, MediaProcessorDirection direction) argument 841 UnregisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor, MediaProcessorDirection direction) argument [all...] |
/external/chromium_org/media/cast/net/ |
H A D | cast_transport_sender_impl.cc | 141 pacer_.RegisterAudioSsrc(config.ssrc); 142 pacer_.RegisterPrioritySsrc(config.ssrc); 152 weak_factory_.GetWeakPtr(), config.ssrc, 159 config.ssrc, 161 pacer_.RegisterAudioSsrc(config.ssrc); 185 weak_factory_.GetWeakPtr(), config.ssrc, 192 config.ssrc, 194 pacer_.RegisterVideoSsrc(config.ssrc); 217 void CastTransportSenderImpl::InsertFrame(uint32 ssrc, argument 219 if (audio_sender_ && ssrc 228 SendSenderReport( uint32 ssrc, base::TimeTicks current_time, uint32 current_time_as_rtp_timestamp) argument 245 CancelSendingFrames( uint32 ssrc, const std::vector<uint32>& frame_ids) argument 257 ResendFrameForKickstart(uint32 ssrc, uint32 frame_id) argument 274 ResendPackets( uint32 ssrc, const MissingFramesAndPacketsMap& missing_packets, bool cancel_rtx_if_not_in_list, const DedupInfo& dedup_info) argument 365 OnReceivedCastMessage( uint32 ssrc, const RtcpCastMessageCallback& cast_message_cb, const RtcpCastMessage& cast_message) argument [all...] |
/external/chromium_org/third_party/libsrtp/srtp/include/ |
H A D | srtp_priv.h | 80 uint32_t ssrc; /* synchronization source */ member in struct:__anon12837 94 uint32_t ssrc; /* synchronization source */ member in struct:__anon12838 120 uint32_t ssrc; /* synchronization source */ member in struct:__anon12840 139 uint32_t ssrc; /* synchronization source */ member in struct:__anon12842 165 * srtp_get_stream(ssrc) returns a pointer to the stream corresponding 166 * to ssrc, or NULL if no stream exists for that ssrc 170 srtp_get_stream(srtp_t srtp, uint32_t ssrc); 210 uint32_t ssrc; member in struct:srtp_stream_ctx_t
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/external/srtp/include/ |
H A D | srtp_priv.h | 80 uint32_t ssrc; /* synchronization source */ member in struct:__anon31217 94 uint32_t ssrc; /* synchronization source */ member in struct:__anon31218 120 uint32_t ssrc; /* synchronization source */ member in struct:__anon31220 139 uint32_t ssrc; /* synchronization source */ member in struct:__anon31222 165 * srtp_get_stream(ssrc) returns a pointer to the stream corresponding 166 * to ssrc, or NULL if no stream exists for that ssrc 170 srtp_get_stream(srtp_t srtp, uint32_t ssrc); 217 uint32_t ssrc; member in struct:srtp_stream_ctx_t
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