| /external/webrtc/logging/rtc_event_log/events/ |
| D | rtc_event_audio_playout.h | 38 const uint32_t ssrc_; variable
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| /external/webrtc/video/ |
| D | rtp_video_stream_receiver_frame_transformer_delegate.h | 69 const uint32_t ssrc_; variable
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| D | rtp_video_stream_receiver_frame_transformer_delegate.cc | 65 const uint32_t ssrc_; member in webrtc::__anon3ca851090111::TransformableVideoReceiverFrame
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| /external/webrtc/pc/ |
| D | jitter_buffer_delay.h | 45 absl::optional<uint32_t> ssrc_; variable
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| D | video_rtp_receiver.h | 129 absl::optional<uint32_t> ssrc_; variable
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| D | audio_rtp_receiver.h | 121 absl::optional<uint32_t> ssrc_; variable
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| D | rtp_sender.h | 181 uint32_t ssrc_ = 0; variable
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_video_frame_transformer_delegate.h | 70 const uint32_t ssrc_; variable
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| D | receive_statistics_impl.h | 66 const uint32_t ssrc_; variable
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| D | fec_test_helper.h | 53 uint32_t ssrc_; variable
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| D | rtp_sender.h | 186 const uint32_t ssrc_; variable
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| D | rtp_sender_egress.h | 131 const uint32_t ssrc_; variable
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| D | forward_error_correction.h | 334 const uint32_t ssrc_; variable
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| D | rtcp_sender.h | 215 const uint32_t ssrc_; variable
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| D | rtp_sender_video_frame_transformer_delegate.cc | 88 const uint32_t ssrc_; member in webrtc::__anon73054be90111::TransformableVideoSenderFrame
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| /external/webrtc/call/ |
| D | rtp_payload_params.h | 113 const uint32_t ssrc_; variable
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| /external/webrtc/test/scenario/ |
| D | audio_stream.h | 47 uint32_t ssrc_; variable
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | rtp_generator.h | 50 const uint32_t ssrc_; variable
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| /external/webrtc/modules/rtp_rtcp/include/ |
| D | flexfec_sender.h | 84 const uint32_t ssrc_; variable
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| /external/webrtc/modules/rtp_rtcp/source/deprecated/ |
| D | deprecated_rtp_sender_egress.h | 106 const uint32_t ssrc_; variable
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| /external/webrtc/video/end_to_end_tests/ |
| D | multi_stream_tests.cc | 52 const uint32_t ssrc_; in TEST() member in webrtc::TEST::VideoOutputObserver
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| /external/webrtc/audio/ |
| D | channel_receive_frame_transformer_delegate.cc | 43 uint32_t ssrc_; member in webrtc::__anon9a6b5ad40111::TransformableAudioFrame
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| D | channel_send_frame_transformer_delegate.cc | 59 uint32_t ssrc_; member in webrtc::__anon9467965b0111::TransformableAudioFrame
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| /external/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_unittest_helper.h | 101 uint32_t ssrc_; variable
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| /external/openscreen/cast/streaming/ |
| D | rtp_packetizer_unittest.cc | 129 const Ssrc ssrc_{GenerateSsrc(true)}; member in openscreen::cast::__anoned8754e60111::RtpPacketizerTest
|