| /external/chromium_org/media/cast/net/rtp/ |
| rtp_packet_builder.cc | 27 ssrc_(0) {}
56 void RtpPacketBuilder::SetSsrc(uint32 ssrc) { ssrc_ = ssrc; }
90 big_endian_writer.WriteU32(ssrc_);
|
| rtp_packet_builder.h | 41 uint32 ssrc_; member in class:media::cast::RtpPacketBuilder
|
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| testutils.h | 156 ScreencastEventCatcher() : ssrc_(0), ev_(rtc::WE_RESIZE) { }
157 uint32 ssrc() const { return ssrc_; }
160 ssrc_ = ssrc;
164 uint32 ssrc_; member in class:cricket::ScreencastEventCatcher
170 VideoMediaErrorCatcher() : ssrc_(0), error_(VideoMediaChannel::ERROR_NONE) { }
171 uint32 ssrc() const { return ssrc_; }
174 ssrc_ = ssrc;
178 uint32 ssrc_; member in class:cricket::VideoMediaErrorCatcher
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_generator.h | 32 ssrc_(ssrc),
52 const uint32_t ssrc_; member in class:webrtc::test::RtpGenerator
|
| rtp_generator.cc | 30 rtp_header->header.ssrc = ssrc_;
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| receive_statistics_unittest.cc | 139 : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {}
144 ssrc_ = ssrc;
150 uint32_t ssrc_; member in class:webrtc::TestCallback
183 EXPECT_EQ(callback.ssrc_, kSsrc1);
225 : StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {}
230 ssrc_ = ssrc;
243 EXPECT_EQ(ssrc, ssrc_);
252 uint32_t ssrc_; member in class:webrtc::RtpTestCallback
|
| rtp_receiver_impl.cc | 78 ssrc_(0),
143 return ssrc_;
193 cb_rtp_feedback_->ResetStatistics(ssrc_);
277 if (ssrc_ != rtp_header.ssrc ||
278 (last_received_payload_type == -1 && ssrc_ == 0)) {
282 cb_rtp_feedback_->ResetStatistics(ssrc_);
289 if (ssrc_ != 0) {
308 ssrc_ = rtp_header.ssrc;
|
| rtp_receiver_impl.h | 94 uint32_t ssrc_; member in class:webrtc::RtpReceiverImpl
|
| rtp_sender.cc | 104 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
122 ssrc_db_.ReturnSSRC(ssrc_);
414 ssrc = ssrc_;
543 ssrc = ssrc_;
972 ssrc = ssrc_;
[all...] |
| rtp_sender_unittest.cc | 818 uint32_t ssrc_; member in class:webrtc::TestCallback
869 uint32_t ssrc_; member in class:webrtc::TestCallback
947 uint32_t ssrc_; member in class:webrtc::TestCallback
[all...] |
| receive_statistics_impl.cc | 34 ssrc_(0),
81 ssrc_ = header.ssrc;
181 ssrc = ssrc_;
192 ssrc = ssrc_;
|
| receive_statistics_impl.h | 65 uint32_t ssrc_; member in class:webrtc::StreamStatisticianImpl
|
| rtp_fec_unittest.cc | 44 : fec_(new ForwardErrorCorrection()), ssrc_(rand()), fec_seq_num_(0) {}
47 int ssrc_; member in class:RtpFecTest
[all...] |
| rtcp_packet.h | 430 ssrc_(0) {
437 ssrc_ = ssrc;
463 uint32_t ssrc_; member in class:webrtc::rtcp::App
[all...] |
| /external/chromium_org/media/cast/sender/ |
| frame_sender.cc | 23 #define SENDER_SSRC (is_audio_ ? "AUDIO[" : "VIDEO[") << ssrc_ << "] "
37 ssrc_(ssrc),
96 transport_sender_->SendSenderReport(ssrc_, now, now_as_rtp_timestamp);
160 transport_sender_->ResendFrameForKickstart(ssrc_, last_sent_frame_id_);
263 transport_sender_->InsertFrame(ssrc_, *encoded_frame);
333 transport_sender_->CancelSendingFrames(ssrc_, cancel_sending_frames);
|
| frame_sender.h | 78 const uint32 ssrc_; member in class:media::cast::FrameSender
|
| /external/chromium_org/media/cast/net/rtcp/ |
| rtcp_builder.cc | 148 ssrc_(sending_ssrc),
224 writer_.WriteU32(ssrc_);
252 writer_.WriteU32(ssrc_); // Add our own SSRC.
266 writer_.WriteU32(ssrc_); // Add our own SSRC.
332 writer_.WriteU32(ssrc_);
359 writer_.WriteU32(ssrc_); // Add our own SSRC.
363 writer_.WriteU32(ssrc_); // Add the media (received RTP) SSRC.
380 writer_.WriteU32(ssrc_); // Add our own SSRC.
|
| rtcp_builder.h | 79 const uint32 ssrc_; member in class:media::cast::RtcpBuilder
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| test_util.h | 94 uint32_t ssrc_; member in class:FileOutputFrameReceiver
|
| rtp_player.cc | 49 ssrc_(ssrc),
59 uint32_t ssrc() const { return ssrc_; }
66 uint32_t ssrc_; member in class:webrtc::rtpplayer::RawRtpPacket
278 ssrc_(ssrc),
289 lost_packets_->SetResendTime(ssrc_, sequence_numbers[i]);
293 virtual uint32_t ssrc() const { return ssrc_; }
304 uint32_t ssrc_; member in class:webrtc::rtpplayer::SsrcHandlers::Handler
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| mediastreamhandler.h | 58 uint32 ssrc() const { return ssrc_; }
66 uint32 ssrc_; member in class:webrtc::TrackHandler
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| srtpfilter_unittest.cc | 732 : ssrc_(0U),
742 ssrc_ = ssrc;
747 ssrc_ = 0U;
753 uint32 ssrc_; member in class:SrtpStatTest
[all...] |
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.h | 96 unsigned int ssrc_; member in class:webrtc::testing::RtpStream
|
| remote_bitrate_estimator_unittest_helper.cc | 36 ssrc_(ssrc),
67 packet->ssrc = ssrc_;
91 rtcp->ssrc = ssrc_;
106 return ssrc_;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_impl.h | 381 uint32_t ssrc_ GUARDED_BY(crit_sect_);
|
