| /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; }
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| rtpfb.h | 30 void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
31 void To(uint32_t ssrc) { media_ssrc_ = ssrc; }
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| 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);
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| voip_metric.h | 38 void To(uint32_t ssrc) { ssrc_ = ssrc; }
43 uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtcp::VoipMetric
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| tmmbr.h | 31 void From(uint32_t ssrc) {
32 tmmbr_.SenderSSRC = ssrc;
34 void To(uint32_t ssrc) {
35 tmmbr_item_.SSRC = ssrc;
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| 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);
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| /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)
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| rtpreceiver.cc | 35 uint32_t ssrc,
39 ssrc_(ssrc),
86 uint32_t ssrc,
88 : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
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| /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);
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| 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
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| 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.
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| 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.
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| /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:__anon22821
94 uint32_t ssrc; /* synchronization source */ member in struct:__anon22822
120 uint32_t ssrc; /* synchronization source */ member in struct:__anon22824
139 uint32_t ssrc; /* synchronization source */ member in struct:__anon22826
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,
|
