| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| D | testutils.h | 156 ScreencastEventCatcher() : ssrc_(0), ev_(rtc::WE_RESIZE) { } in ScreencastEventCatcher()
157 uint32 ssrc() const { return ssrc_; } in ssrc()
160 ssrc_ = ssrc; in OnEvent()
164 uint32 ssrc_;
170 VideoMediaErrorCatcher() : ssrc_(0), error_(VideoMediaChannel::ERROR_NONE) { } in VideoMediaErrorCatcher()
171 uint32 ssrc() const { return ssrc_; } in ssrc()
174 ssrc_ = ssrc; in OnError()
178 uint32 ssrc_;
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | receive_statistics_unittest.cc | 139 : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {} in TEST_F()
144 ssrc_ = ssrc; in TEST_F()
150 uint32_t ssrc_; in TEST_F() member in webrtc::TEST_F::TestCallback
183 EXPECT_EQ(callback.ssrc_, kSsrc1); in TEST_F()
225 : StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {} in RtpTestCallback()
230 ssrc_ = ssrc; in DataCountersUpdated()
243 EXPECT_EQ(ssrc, ssrc_); in ExpectMatches()
252 uint32_t ssrc_; member in webrtc::RtpTestCallback
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| D | rtp_receiver_impl.cc | 78 ssrc_(0), in RtpReceiverImpl()
143 return ssrc_; in SSRC()
193 cb_rtp_feedback_->ResetStatistics(ssrc_); in IncomingRtpPacket()
277 if (ssrc_ != rtp_header.ssrc || in CheckSSRCChanged()
278 (last_received_payload_type == -1 && ssrc_ == 0)) { in CheckSSRCChanged()
282 cb_rtp_feedback_->ResetStatistics(ssrc_); in CheckSSRCChanged()
289 if (ssrc_ != 0) { in CheckSSRCChanged()
308 ssrc_ = rtp_header.ssrc; in CheckSSRCChanged()
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| D | rtp_sender.cc | 104 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0. in RTPSender()
122 ssrc_db_.ReturnSSRC(ssrc_); in ~RTPSender()
414 ssrc = ssrc_; in SendOutgoingData()
543 ssrc = ssrc_; in SendPadData()
972 ssrc = ssrc_; in UpdateDelayStatistics()
1019 ssrc = ssrc_; in ResetDataCounters()
1103 return CreateRTPHeader(data_buffer, payload_type, ssrc_, marker_bit, in BuildRTPheader()
1412 ssrc_db_.ReturnSSRC(ssrc_); in SetSendingStatus()
1413 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0. in SetSendingStatus()
1463 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0. in GenerateNewSSRC()
[all …]
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| D | rtp_sender_unittest.cc | 796 : FrameCountObserver(), num_calls_(0), ssrc_(0), in TEST_F()
804 ssrc_ = ssrc; in TEST_F()
818 uint32_t ssrc_; in TEST_F() member in webrtc::TEST_F::TestCallback
839 EXPECT_EQ(ssrc, callback.ssrc_); in TEST_F()
848 EXPECT_EQ(ssrc, callback.ssrc_); in TEST_F()
859 : BitrateStatisticsObserver(), num_calls_(0), ssrc_(0), bitrate_() {} in TEST_F()
864 ssrc_ = ssrc; in TEST_F()
869 uint32_t ssrc_; in TEST_F() member in webrtc::TEST_F::TestCallback
912 EXPECT_EQ(ssrc, callback.ssrc_); in TEST_F()
938 : StreamDataCountersCallback(), ssrc_(0), counters_() {} in TEST_F()
[all …]
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| D | receive_statistics_impl.cc | 34 ssrc_(0), in StreamStatisticianImpl()
81 ssrc_ = header.ssrc; in UpdateCounters()
181 ssrc = ssrc_; in NotifyRtpCallback()
192 ssrc = ssrc_; in NotifyRtcpCallback()
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| D | rtp_fec_unittest.cc | 44 : fec_(new ForwardErrorCorrection()), ssrc_(rand()), fec_seq_num_(0) {} in RtpFecTest()
47 int ssrc_; member in RtpFecTest
878 received_packet->ssrc = ssrc_; in ReceivedPackets()
928 webrtc::RtpUtility::AssignUWord32ToBuffer(&media_packet->data[8], ssrc_); in ConstructMediaPacketsSeqNum()
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| D | rtp_receiver_impl.h | 94 uint32_t ssrc_; variable
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| D | receive_statistics_impl.h | 65 uint32_t ssrc_; variable
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| D | rtcp_packet.h | 430 ssrc_(0) { in App()
437 ssrc_ = ssrc; in From()
463 uint32_t ssrc_; variable
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| D | rtcp_receiver_unittest.cc | 817 RtcpCallbackImpl() : RtcpStatisticsCallback(), ssrc_(0) {} in TEST_F()
823 ssrc_ = ssrc; in TEST_F()
828 return ssrc_ == ssrc && in TEST_F()
836 uint32_t ssrc_; in TEST_F() member in webrtc::__anoncc59f3960111::TEST_F::RtcpCallbackImpl
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| /external/chromium_org/media/cast/net/rtp/ |
| D | rtp_packet_builder.cc | 27 ssrc_(0) {} in RtpPacketBuilder()
56 void RtpPacketBuilder::SetSsrc(uint32 ssrc) { ssrc_ = ssrc; } in SetSsrc()
90 big_endian_writer.WriteU32(ssrc_); in BuildCommonHeader()
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| D | rtp_packet_builder.h | 41 uint32 ssrc_; variable
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| /external/chromium_org/media/cast/sender/ |
| D | frame_sender.cc | 23 #define SENDER_SSRC (is_audio_ ? "AUDIO[" : "VIDEO[") << ssrc_ << "] "
37 ssrc_(ssrc), in FrameSender()
96 transport_sender_->SendSenderReport(ssrc_, now, now_as_rtp_timestamp); in SendRtcpReport()
160 transport_sender_->ResendFrameForKickstart(ssrc_, last_sent_frame_id_); in ResendForKickstart()
263 transport_sender_->InsertFrame(ssrc_, *encoded_frame); in SendEncodedFrame()
333 transport_sender_->CancelSendingFrames(ssrc_, cancel_sending_frames); in OnReceivedCastFeedback()
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| D | frame_sender.h | 78 const uint32 ssrc_; variable
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| /external/chromium_org/media/cast/net/rtcp/ |
| D | rtcp_builder.cc | 148 ssrc_(sending_ssrc), in RtcpBuilder()
224 writer_.WriteU32(ssrc_); in AddRR()
252 writer_.WriteU32(ssrc_); // Add our own SSRC. in AddRrtr()
266 writer_.WriteU32(ssrc_); // Add our own SSRC. in AddCast()
332 writer_.WriteU32(ssrc_); in AddSR()
359 writer_.WriteU32(ssrc_); // Add our own SSRC. in AddDlrrRb()
363 writer_.WriteU32(ssrc_); // Add the media (received RTP) SSRC. in AddDlrrRb()
380 writer_.WriteU32(ssrc_); // Add our own SSRC. in AddReceiverLog()
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| D | rtcp_builder.h | 79 const uint32 ssrc_; variable
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| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| D | srtpfilter_unittest.cc | 732 : ssrc_(0U), in SrtpStatTest()
742 ssrc_ = ssrc; in OnSrtpError()
747 ssrc_ = 0U; in Reset()
753 uint32 ssrc_; member in SrtpStatTest
764 EXPECT_EQ(0U, ssrc_); in TEST_F()
769 EXPECT_EQ(1U, ssrc_); in TEST_F()
774 EXPECT_EQ(1U, ssrc_); in TEST_F()
780 EXPECT_EQ(0U, ssrc_); in TEST_F()
787 EXPECT_EQ(1U, ssrc_); in TEST_F()
795 EXPECT_EQ(0U, ssrc_); in TEST_F()
[all …]
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| D | rtp_player.cc | 49 ssrc_(ssrc), in RawRtpPacket()
59 uint32_t ssrc() const { return ssrc_; } in ssrc()
66 uint32_t ssrc_; member in webrtc::rtpplayer::RawRtpPacket
278 ssrc_(ssrc), in Handler()
289 lost_packets_->SetResendTime(ssrc_, sequence_numbers[i]); in ResendPackets()
293 virtual uint32_t ssrc() const { return ssrc_; } in ssrc()
304 uint32_t ssrc_; member in webrtc::rtpplayer::SsrcHandlers::Handler
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| D | test_util.h | 94 uint32_t ssrc_; variable
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| D | rtp_generator.h | 32 ssrc_(ssrc), in seq_number_()
52 const uint32_t ssrc_; variable
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| D | rtp_generator.cc | 30 rtp_header->header.ssrc = ssrc_; in GetRtpHeader()
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| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_unittest_helper.cc | 36 ssrc_(ssrc), in RtpStream()
67 packet->ssrc = ssrc_; in GenerateFrame()
91 rtcp->ssrc = ssrc_; in Rtcp()
106 return ssrc_; in ssrc()
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| D | remote_bitrate_estimator_unittest_helper.h | 96 unsigned int ssrc_; variable
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| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| D | mediastreamhandler.h | 58 uint32 ssrc() const { return ssrc_; } in ssrc()
66 uint32 ssrc_; variable
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