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Searched refs:ssrc_ (Results 1 – 25 of 35) sorted by relevance

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/external/webrtc/talk/app/webrtc/
Drtpsender.cc119 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); in SetTrack()
135 stats_->AddLocalAudioTrack(track_.get(), ssrc_); in SetTrack()
139 provider_->SetAudioSend(ssrc_, false, options, nullptr); in SetTrack()
145 if (stopped_ || ssrc == ssrc_) { in SetSsrc()
151 provider_->SetAudioSend(ssrc_, false, options, nullptr); in SetSsrc()
153 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); in SetSsrc()
156 ssrc_ = ssrc; in SetSsrc()
160 stats_->AddLocalAudioTrack(track_.get(), ssrc_); in SetSsrc()
176 provider_->SetAudioSend(ssrc_, false, options, nullptr); in Stop()
178 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); in Stop()
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Drtpreceiver.cc39 ssrc_(ssrc), in AudioRtpReceiver()
66 provider_->SetAudioPlayoutVolume(ssrc_, volume); in OnSetVolume()
74 provider_->SetAudioPlayout(ssrc_, false); in Stop()
82 provider_->SetAudioPlayout(ssrc_, track_->enabled()); in Reconfigure()
88 : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) { in VideoRtpReceiver()
90 provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput()); in VideoRtpReceiver()
104 provider_->SetVideoPlayout(ssrc_, false, nullptr); in Stop()
Drtpsender.h97 uint32_t ssrc() const override { return ssrc_; } in ssrc()
113 bool can_send_track() const { return track_ && ssrc_; } in can_send_track()
123 uint32_t ssrc_ = 0; variable
155 uint32_t ssrc() const override { return ssrc_; } in ssrc()
171 bool can_send_track() const { return track_ && ssrc_; } in can_send_track()
180 uint32_t ssrc_ = 0; variable
Drtpreceiver.h73 const uint32_t ssrc_; variable
98 uint32_t ssrc_; variable
/external/webrtc/talk/media/base/
Dtestutils.h160 ScreencastEventCatcher() : ssrc_(0), ev_(rtc::WE_RESIZE) { } in ScreencastEventCatcher()
161 uint32_t ssrc() const { return ssrc_; } in ssrc()
164 ssrc_ = ssrc; in OnEvent()
168 uint32_t ssrc_;
174 VideoMediaErrorCatcher() : ssrc_(0), error_(VideoMediaChannel::ERROR_NONE) { } in VideoMediaErrorCatcher()
175 uint32_t ssrc() const { return ssrc_; } in ssrc()
178 ssrc_ = ssrc; in OnError()
182 uint32_t ssrc_;
/external/webrtc/webrtc/modules/rtp_rtcp/source/
Drtcp_sender.cc161 ssrc_(0), in RTCPSender()
302 if (ssrc_ != 0) { in SetSSRC()
308 ssrc_ = ssrc; in SetSSRC()
480 report->From(ssrc_); in BuildSR()
501 sdes->WithCName(ssrc_, cname_); in BuildSDES()
511 report->From(ssrc_); in BuildRR()
521 pli->From(ssrc_); in BuildPLI()
528 ssrc_, packet_type_counter_.pli_packets); in BuildPLI()
538 fir->From(ssrc_); in BuildFIR()
546 ssrc_, packet_type_counter_.fir_packets); in BuildFIR()
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Dreceive_statistics_unittest.cc156 : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {} in TEST_F()
161 ssrc_ = ssrc; in TEST_F()
169 uint32_t ssrc_; in TEST_F() member in webrtc::TEST_F::TestCallback
202 EXPECT_EQ(callback.ssrc_, kSsrc1); in TEST_F()
244 : StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {} in RtpTestCallback()
249 ssrc_ = ssrc; in DataCountersUpdated()
266 EXPECT_EQ(ssrc, ssrc_); in Matches()
273 uint32_t ssrc_; member in webrtc::RtpTestCallback
Drtp_sender.cc70 ssrc_(0) {} in BitrateAggregator()
76 ssrc_); in OnStatsUpdated()
86 void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; } in set_ssrc()
112 uint32_t ssrc_; member in webrtc::BitrateAggregator
185 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0. in RTPSender()
187 bitrates_->set_ssrc(ssrc_); in RTPSender()
197 ssrc_db_.ReturnSSRC(ssrc_); in ~RTPSender()
513 ssrc = ssrc_; in SendOutgoingData()
635 ssrc = ssrc_; in SendPadData()
1103 ssrc = ssrc_; in UpdateDelayStatistics()
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Drtp_receiver_impl.cc76 ssrc_(0), in RtpReceiverImpl()
140 return ssrc_; in SSRC()
263 if (ssrc_ != rtp_header.ssrc || in CheckSSRCChanged()
264 (last_received_payload_type == -1 && ssrc_ == 0)) { in CheckSSRCChanged()
273 if (ssrc_ != 0) { in CheckSSRCChanged()
291 ssrc_ = rtp_header.ssrc; in CheckSSRCChanged()
Drtp_sender_unittest.cc979 TestCallback() : FrameCountObserver(), num_calls_(0), ssrc_(0) {} in TEST_F()
985 ssrc_ = ssrc; in TEST_F()
990 uint32_t ssrc_; in TEST_F() member in webrtc::TEST_F::TestCallback
1011 EXPECT_EQ(ssrc, callback.ssrc_); in TEST_F()
1020 EXPECT_EQ(ssrc, callback.ssrc_); in TEST_F()
1030 TestCallback() : BitrateStatisticsObserver(), num_calls_(0), ssrc_(0) {} in TEST_F()
1037 ssrc_ = ssrc; in TEST_F()
1043 uint32_t ssrc_; in TEST_F() member in webrtc::TEST_F::TestCallback
1084 EXPECT_EQ(ssrc, callback.ssrc_); in TEST_F()
1110 TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {} in TEST_F()
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Dreceive_statistics_impl.cc34 ssrc_(0), in StreamStatisticianImpl()
64 ssrc_ = header.ssrc; in UpdateCounters()
155 ssrc = ssrc_; in NotifyRtpCallback()
166 ssrc = ssrc_; in NotifyRtcpCallback()
Drtp_fec_unittest.cc47 ssrc_(random_.Rand<uint32_t>()), in RtpFecTest()
52 int ssrc_; member in RtpFecTest
883 received_packet->ssrc = ssrc_; in ReceivedPackets()
931 webrtc::ByteWriter<uint32_t>::WriteBigEndian(&media_packet->data[8], ssrc_); in ConstructMediaPacketsSeqNum()
Drtp_receiver_impl.h88 uint32_t ssrc_; variable
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/
Dapp.h28 App() : sub_type_(0), ssrc_(0), name_(0) {} in App()
36 void From(uint32_t ssrc) { ssrc_ = ssrc; } in From()
42 uint32_t ssrc() const { return ssrc_; } in ssrc()
57 uint32_t ssrc_; variable
Dvoip_metric.h38 void To(uint32_t ssrc) { ssrc_ = ssrc; } in To()
43 uint32_t ssrc() const { return ssrc_; } in ssrc()
47 uint32_t ssrc_;
Dvoip_metric.cc41 VoipMetric::VoipMetric() : ssrc_(0) { in VoipMetric()
49 ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]); in Parse()
80 ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ssrc_); in Create()
Dapp.cc39 ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]); in Parse()
70 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_); in Create()
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
Dpacket_source.h26 PacketSource() : use_ssrc_filter_(false), ssrc_(0) {} in PacketSource()
39 ssrc_ = ssrc; in SelectSsrc()
46 uint32_t ssrc_; // The selected SSRC. All other SSRCs will be discarded. variable
Drtp_generator.h32 ssrc_(ssrc), in seq_number_()
52 const uint32_t ssrc_; variable
Drtp_generator.cc30 rtp_header->header.ssrc = ssrc_; in GetRtpHeader()
Drtp_file_source.cc77 (use_ssrc_filter_ && packet->header().ssrc != ssrc_)) { in NextPacket()
/external/webrtc/webrtc/modules/video_coding/test/
Drtp_player.cc51 ssrc_(ssrc), in RawRtpPacket()
61 uint32_t ssrc() const { return ssrc_; } in ssrc()
68 uint32_t ssrc_; member in webrtc::rtpplayer::RawRtpPacket
277 ssrc_(ssrc), in Handler()
288 lost_packets_->SetResendTime(ssrc_, sequence_numbers[i]); in ResendPackets()
292 virtual uint32_t ssrc() const { return ssrc_; } in ssrc()
301 uint32_t ssrc_; member in webrtc::rtpplayer::SsrcHandlers::Handler
/external/webrtc/talk/session/media/
Dsrtpfilter_unittest.cc763 : ssrc_(0U), in SrtpStatTest()
774 ssrc_ = ssrc; in OnSrtpError()
779 ssrc_ = 0U; in Reset()
785 uint32_t ssrc_; member in SrtpStatTest
796 EXPECT_EQ(0U, ssrc_); in TEST_F()
801 EXPECT_EQ(1U, ssrc_); in TEST_F()
806 EXPECT_EQ(1U, ssrc_); in TEST_F()
812 EXPECT_EQ(0U, ssrc_); in TEST_F()
819 EXPECT_EQ(1U, ssrc_); in TEST_F()
827 EXPECT_EQ(0U, ssrc_); in TEST_F()
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/
Dremote_bitrate_estimator_unittest_helper.cc38 ssrc_(ssrc), in RtpStream()
69 packet->ssrc = ssrc_; in GenerateFrame()
93 rtcp->ssrc = ssrc_; in Rtcp()
108 return ssrc_; in ssrc()
Dremote_bitrate_estimator_unittest_helper.h97 unsigned int ssrc_; variable

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