/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
psfb.h | 30 void From(uint32_t ssrc) { sender_ssrc_ = ssrc; } 31 void To(uint32_t ssrc) { media_ssrc_ = ssrc; }
|
rtpfb.h | 30 void From(uint32_t ssrc) { sender_ssrc_ = ssrc; } 31 void To(uint32_t ssrc) { media_ssrc_ = ssrc; }
|
tmmbn.h | 32 void From(uint32_t ssrc) { 33 tmmbn_.SenderSSRC = ssrc; 36 bool WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead);
|
voip_metric.h | 38 void To(uint32_t ssrc) { ssrc_ = ssrc; } 43 uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtcp::VoipMetric
|
tmmbr.h | 31 void From(uint32_t ssrc) { 32 tmmbr_.SenderSSRC = ssrc; 34 void To(uint32_t ssrc) { 35 tmmbr_item_.SSRC = ssrc;
|
dlrr.h | 27 uint32_t ssrc; member in struct:webrtc::rtcp::Dlrr::SubBlock 51 bool WithDlrrItem(uint32_t ssrc, uint32_t last_rr, uint32_t delay_last_rr);
|
/external/webrtc/talk/app/webrtc/ |
mediastreamprovider.h | 48 // TODO(deadbeef): Change the key from an ssrc to a "sender_id" or 60 // Enable/disable the audio playout of a remote audio track with |ssrc|. 61 virtual void SetAudioPlayout(uint32_t ssrc, bool enable) = 0; 62 // Enable/disable sending audio on the local audio track with |ssrc|. 64 virtual void SetAudioSend(uint32_t ssrc, 69 // Sets the audio playout volume of a remote audio track with |ssrc|. 71 virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0; 74 // Only one audio sink is supported per ssrc and ownership of the sink is 77 uint32_t ssrc, 88 virtual bool SetCaptureDevice(uint32_t ssrc, [all...] |
rtpsenderinterface.h | 51 // Used to set the SSRC of the sender, once a local description has been set. 52 // If |ssrc| is 0, this indiates that the sender should disconnect from the 55 virtual void SetSsrc(uint32_t ssrc) = 0; 56 virtual uint32_t ssrc() const = 0; 80 PROXY_CONSTMETHOD0(uint32_t, ssrc)
|
rtpreceiver.cc | 35 uint32_t ssrc, 39 ssrc_(ssrc), 86 uint32_t ssrc, 88 : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
|
/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
ssrc_database.cc | 35 while (true) { // Try until get a new ssrc. 36 // 0 and 0xffffffff are invalid values for SSRC. 37 uint32_t ssrc = random_.Rand(1u, 0xfffffffe); local 38 if (ssrcs_.insert(ssrc).second) { 39 return ssrc; 44 void SSRCDatabase::RegisterSSRC(uint32_t ssrc) { 46 ssrcs_.insert(ssrc); 49 void SSRCDatabase::ReturnSSRC(uint32_t ssrc) { 51 ssrcs_.erase(ssrc);
|
ssrc_database.h | 30 void RegisterSSRC(uint32_t ssrc); 31 void ReturnSSRC(uint32_t ssrc); 40 // Friend function to allow the SSRC destructor to be accessed from the
|
rtcp_packet.h | 136 // | SSRC of sender | 159 void From(uint32_t ssrc) { 160 sr_.SenderSSRC = ssrc; 208 // chunk | SSRC/CSRC_1 | 213 // chunk | SSRC/CSRC_2 | 233 bool WithCName(uint32_t ssrc, const std::string& cname); 236 uint32_t ssrc; member in struct:webrtc::rtcp::Sdes::Chunk 279 void From(uint32_t ssrc) { 280 rpsi_.SenderSSRC = ssrc; 282 void To(uint32_t ssrc) { [all...] |
/external/webrtc/webrtc/video/ |
encoder_state_feedback.cc | 31 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) { 32 owner_->OnReceivedIntraFrameRequest(ssrc); 34 virtual void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) { 35 owner_->OnReceivedSLI(ssrc, picture_id); 37 virtual void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) { 38 owner_->OnReceivedRPSI(ssrc, picture_id); 61 for (uint32_t ssrc : ssrcs) { 62 RTC_DCHECK(encoders_.find(ssrc) == encoders_.end()); 63 encoders_[ssrc] = encoder; 83 void EncoderStateFeedback::OnReceivedIntraFrameRequest(uint32_t ssrc) { [all...] |
vie_remb_unittest.cc | 52 unsigned int ssrc = 1234; local 53 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 77 unsigned int ssrc = 1234; local 78 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 103 unsigned int ssrc[] = { 1234, 5678 }; local 104 std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0])) 134 unsigned int ssrc[] = { 1234, 5678 }; local 168 unsigned int ssrc[] = { 1234, 5678 }; local 202 unsigned int ssrc = 1234; local 235 unsigned int ssrc = 1234; local [all...] |
encoder_state_feedback.h | 39 void AddEncoder(const std::vector<uint32_t>& ssrc, ViEEncoder* encoder); 50 void OnReceivedIntraFrameRequest(uint32_t ssrc); 51 void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id); 52 void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id); 63 // Maps a unique ssrc to the given encoder.
|
send_statistics_proxy_unittest.cc | 104 const uint32_t ssrc = *it; local 105 VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc]; 108 uint32_t offset = ssrc * sizeof(RtcpStatistics); 113 callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc); 118 const uint32_t ssrc = *it; local 119 VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc]; 122 uint32_t offset = ssrc * sizeof(RtcpStatistics); 127 callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc); 162 const uint32_t ssrc = *it; local 164 VideoSendStream::StreamStats& stats = expected_.substreams[ssrc]; 175 const uint32_t ssrc = *it; local 195 const uint32_t ssrc = *it; local 211 const uint32_t ssrc = *it; local 234 const uint32_t ssrc = *it; local 247 const uint32_t ssrc = *it; local 267 const uint32_t ssrc = *it; local 279 const uint32_t ssrc = *it; local [all...] |
receive_statistics_proxy.cc | 23 ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock) 30 stats_.ssrc = ssrc; 120 uint32_t ssrc, 123 if (stats_.ssrc != ssrc) 130 uint32_t ssrc) { 134 if (stats_.ssrc != ssrc) 137 report_block_stats_.Store(statistics, ssrc, 0) [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
packet_source.h | 37 virtual void SelectSsrc(uint32_t ssrc) { 39 ssrc_ = ssrc; 44 // If SSRC filtering discards all packet that do not match the SSRC. 45 bool use_ssrc_filter_; // True when SSRC filtering is active. 46 uint32_t ssrc_; // The selected SSRC. All other SSRCs will be discarded.
|
/external/webrtc/webrtc/call/ |
call_unittest.cc | 47 config.rtp.ssrc = 42; 70 for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { 71 config.rtp.ssrc = ssrc; 74 if (ssrc & 1) { 93 for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) [all...] |
/external/webrtc/webrtc/modules/pacing/ |
paced_sender_unittest.cc | 30 bool(uint32_t ssrc, 42 bool TimeToSendPacket(uint32_t ssrc, 70 bool TimeToSendPacket(uint32_t ssrc, 122 uint32_t ssrc, 127 send_bucket_->InsertPacket(priority, ssrc, sequence_number, capture_time_ms, 130 TimeToSendPacket(ssrc, sequence_number, capture_time_ms, false)) 141 uint32_t ssrc = 12345; local 145 ssrc, 151 ssrc, 157 ssrc, 199 uint32_t ssrc = 12345; local 256 uint32_t ssrc = 12345; local 324 uint32_t ssrc = 12345; local 347 uint32_t ssrc = 12345; local 388 uint32_t ssrc = 12345; local 411 uint32_t ssrc = 12345; local 440 uint32_t ssrc = 12346; local 503 uint32_t ssrc = 12346; local 535 uint32_t ssrc = 12346; local 610 uint32_t ssrc = 12346; local 665 uint32_t ssrc = 12346; local 702 uint32_t ssrc = 12346; local 725 uint32_t ssrc = 12346; local 759 uint32_t ssrc = 12346; local 790 uint32_t ssrc = 12346; local 837 uint32_t ssrc = 12346; local 862 uint32_t ssrc = 12346; local [all...] |
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
rtp_rtcp_test.cc | 28 unsigned int SSRC); 33 void SetIncomingSsrc(unsigned int ssrc) { 35 incoming_ssrc_ = ssrc; 44 unsigned int SSRC) { 46 sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel, 47 SSRC); 52 if (incoming_ssrc_ == SSRC) 75 // We'll set up the RTCP CNAME and SSRC to something arbitrary here. 112 unsigned int ssrc; local 113 EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc)); [all...] |
/external/srtp/include/ |
srtp_priv.h | 80 uint32_t ssrc; /* synchronization source */ member in struct:__anon24263 94 uint32_t ssrc; /* synchronization source */ member in struct:__anon24264 120 uint32_t ssrc; /* synchronization source */ member in struct:__anon24266 139 uint32_t ssrc; /* synchronization source */ member in struct:__anon24268 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); 209 * an srtp_stream_t has its own SSRC, encryption key, authentication 217 uint32_t ssrc; member in struct:srtp_stream_ctx_t [all...] |
/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
remote_bitrate_estimator_unittest_helper.h | 54 unsigned int ssrc; member in struct:webrtc::testing::RtpStream::RtpPacket 61 unsigned int ssrc; member in struct:webrtc::testing::RtpStream::RtcpPacket 68 RtpStream(int fps, int bitrate_bps, unsigned int ssrc, unsigned int frequency, 87 unsigned int ssrc() const; 125 // Set the RTP timestamp offset for the stream identified by |ssrc|. 126 void set_rtp_timestamp_offset(unsigned int ssrc, uint32_t offset); 168 void IncomingPacket(uint32_t ssrc, 176 // with a given ssrc. The stream is pushed through a very simple simulated 181 bool GenerateAndProcessFrame(unsigned int ssrc, unsigned int bitrate_bps); 187 unsigned int SteadyStateRun(unsigned int ssrc, [all...] |
/external/webrtc/talk/media/base/ |
fakemediaengine.h | 117 virtual bool RemoveSendStream(uint32_t ssrc) { 118 return RemoveStreamBySsrc(&send_streams_, ssrc); 128 virtual bool RemoveRecvStream(uint32_t ssrc) { 129 return RemoveStreamBySsrc(&receive_streams_, ssrc); 131 bool IsStreamMuted(uint32_t ssrc) const { 132 bool ret = muted_streams_.find(ssrc) != muted_streams_.end(); 133 // If |ssrc = 0| check if the first send stream is muted. 134 if (!ret && ssrc == 0 && !send_streams_.empty()) { 146 bool HasRecvStream(uint32_t ssrc) const { 147 return GetStreamBySsrc(receive_streams_, ssrc) != nullptr 237 uint32_t ssrc; member in struct:cricket::FakeVoiceMediaChannel::DtmfInfo [all...] |
/external/webrtc/webrtc/modules/pacing/mock/ |
mock_paced_sender.h | 27 uint32_t ssrc,
|