| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| packet_source.h | 26 PacketSource() : use_ssrc_filter_(false), ssrc_(0) {}
39 ssrc_ = ssrc;
46 uint32_t ssrc_; // The selected SSRC. All other SSRCs will be discarded. member in class:webrtc::test::PacketSource
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| rtp_generator.h | 32 ssrc_(ssrc),
52 const uint32_t ssrc_; member in class:webrtc::test::RtpGenerator
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| rtp_generator.cc | 30 rtp_header->header.ssrc = ssrc_;
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| rtp_file_source.cc | 77 (use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
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| rtc_event_log_source.cc | 93 !(use_ssrc_filter_ && packet->header().ssrc != ssrc_))
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| /external/webrtc/talk/app/webrtc/ |
| rtpreceiver.cc | 39 ssrc_(ssrc),
66 provider_->SetAudioPlayoutVolume(ssrc_, volume);
74 provider_->SetAudioPlayout(ssrc_, false);
82 provider_->SetAudioPlayout(ssrc_, track_->enabled());
88 : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
90 provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput());
104 provider_->SetVideoPlayout(ssrc_, false, nullptr);
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| rtpsender.cc | 119 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
135 stats_->AddLocalAudioTrack(track_.get(), ssrc_);
139 provider_->SetAudioSend(ssrc_, false, options, nullptr);
145 if (stopped_ || ssrc == ssrc_) {
151 provider_->SetAudioSend(ssrc_, false, options, nullptr);
153 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
156 ssrc_ = ssrc;
160 stats_->AddLocalAudioTrack(track_.get(), ssrc_);
176 provider_->SetAudioSend(ssrc_, false, options, nullptr);
178 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
[all...] |
| rtpsender.h | 97 uint32_t ssrc() const override { return ssrc_; }
113 bool can_send_track() const { return track_ && ssrc_; }
123 uint32_t ssrc_ = 0; member in class:webrtc::AudioRtpSender
155 uint32_t ssrc() const override { return ssrc_; }
171 bool can_send_track() const { return track_ && ssrc_; }
180 uint32_t ssrc_ = 0; member in class:webrtc::VideoRtpSender
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| rtpreceiver.h | 73 const uint32_t ssrc_; member in class:webrtc::AudioRtpReceiver
98 uint32_t ssrc_; member in class:webrtc::VideoRtpReceiver
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| voip_metric.h | 38 void To(uint32_t ssrc) { ssrc_ = ssrc; }
43 uint32_t ssrc() const { return ssrc_; }
47 uint32_t ssrc_; member in class:webrtc::rtcp::VoipMetric
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| app.h | 28 App() : sub_type_(0), ssrc_(0), name_(0) {}
36 void From(uint32_t ssrc) { ssrc_ = ssrc; }
42 uint32_t ssrc() const { return ssrc_; }
57 uint32_t ssrc_; member in class:webrtc::rtcp::App
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| app.cc | 39 ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
70 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_);
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| voip_metric.cc | 41 VoipMetric::VoipMetric() : ssrc_(0) {
49 ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
80 ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ssrc_);
|
| /external/webrtc/talk/media/base/ |
| testutils.h | 160 ScreencastEventCatcher() : ssrc_(0), ev_(rtc::WE_RESIZE) { }
161 uint32_t ssrc() const { return ssrc_; }
164 ssrc_ = ssrc;
168 uint32_t ssrc_; member in class:cricket::ScreencastEventCatcher
174 VideoMediaErrorCatcher() : ssrc_(0), error_(VideoMediaChannel::ERROR_NONE) { }
175 uint32_t ssrc() const { return ssrc_; }
178 ssrc_ = ssrc;
182 uint32_t ssrc_; member in class:cricket::VideoMediaErrorCatcher
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtcp_sender.cc | 161 ssrc_(0),
302 if (ssrc_ != 0) {
308 ssrc_ = ssrc;
480 report->From(ssrc_);
501 sdes->WithCName(ssrc_, cname_);
511 report->From(ssrc_);
521 pli->From(ssrc_);
528 ssrc_, packet_type_counter_.pli_packets);
538 fir->From(ssrc_);
546 ssrc_, packet_type_counter_.fir_packets)
[all...] |
| rtp_receiver_impl.cc | 76 ssrc_(0),
140 return ssrc_;
263 if (ssrc_ != rtp_header.ssrc ||
264 (last_received_payload_type == -1 && ssrc_ == 0)) {
273 if (ssrc_ != 0) {
291 ssrc_ = rtp_header.ssrc;
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| receive_statistics_unittest.cc | 156 : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {}
161 ssrc_ = ssrc;
169 uint32_t ssrc_; member in class:webrtc::TestCallback
202 EXPECT_EQ(callback.ssrc_, kSsrc1);
244 : StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {}
249 ssrc_ = ssrc;
266 EXPECT_EQ(ssrc, ssrc_);
273 uint32_t ssrc_; member in class:webrtc::RtpTestCallback
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| rtp_receiver_impl.h | 88 uint32_t ssrc_; member in class:webrtc::RtpReceiverImpl
|
| rtp_sender.cc | 70 ssrc_(0) {}
76 ssrc_);
86 void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; }
112 uint32_t ssrc_; member in class:webrtc::BitrateAggregator
185 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
187 bitrates_->set_ssrc(ssrc_);
197 ssrc_db_.ReturnSSRC(ssrc_);
513 ssrc = ssrc_;
635 ssrc = ssrc_;
[all...] |
| receive_statistics_impl.cc | 34 ssrc_(0),
64 ssrc_ = header.ssrc;
155 ssrc = ssrc_;
166 ssrc = ssrc_;
|
| receive_statistics_impl.h | 66 uint32_t ssrc_; member in class:webrtc::StreamStatisticianImpl
|
| rtp_sender_unittest.cc | 990 uint32_t ssrc_; member in class:webrtc::TestCallback
1043 uint32_t ssrc_; member in class:webrtc::TestCallback
1119 uint32_t ssrc_; member in class:webrtc::TestCallback
[all...] |
| /external/webrtc/webrtc/modules/video_coding/test/ |
| rtp_player.cc | 51 ssrc_(ssrc),
61 uint32_t ssrc() const { return ssrc_; }
68 uint32_t ssrc_; member in class:webrtc::rtpplayer::RawRtpPacket
277 ssrc_(ssrc),
288 lost_packets_->SetResendTime(ssrc_, sequence_numbers[i]);
292 virtual uint32_t ssrc() const { return ssrc_; }
301 uint32_t ssrc_; member in class:webrtc::rtpplayer::SsrcHandlers::Handler
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| /external/webrtc/talk/session/media/ |
| srtpfilter_unittest.cc | 785 uint32_t ssrc_; member in class:SrtpStatTest
[all...] |
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.h | 97 unsigned int ssrc_; member in class:webrtc::testing::RtpStream
|
