| /external/webrtc/test/ |
| D | call_config_utils_unittest.cc | 30 recv_config.rtp.local_ssrc = 101; in TEST()
53 EXPECT_EQ(recv_config.rtp.local_ssrc, unmarshaled_config.rtp.local_ssrc); in TEST()
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| D | call_config_utils.cc | 41 receive_config.rtp.local_ssrc = json["rtp"]["local_ssrc"].asInt64(); in ParseVideoReceiveStreamJsonConfig()
91 rtp_json["local_ssrc"] = config.rtp.local_ssrc; in GenerateVideoReceiveStreamJsonConfig()
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| /external/webrtc/logging/rtc_event_log/ |
| D | rtc_stream_config.cc | 23 return local_ssrc == other.local_ssrc && remote_ssrc == other.remote_ssrc && in operator ==()
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| D | rtc_stream_config.h | 33 uint32_t local_ssrc = 0; member
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| D | rtc_event_log_parser.cc | 137 streams->insert({conf.config.local_ssrc, {media_type, false}}); in AddSendStreamInfos()
1116 outgoing_video_ssrcs_.insert(video_send_config.config.local_ssrc); in ParseStream()
1124 outgoing_audio_ssrcs_.insert(audio_send_config.config.local_ssrc); in ParseStream()
1386 outgoing_rtp_extensions_maps_[config.value().local_ssrc] = in StoreParsedLegacyEvent()
1411 outgoing_rtp_extensions_maps_[config.value().local_ssrc] = in StoreParsedLegacyEvent()
1676 config.local_ssrc = receiver_config.local_ssrc(); in GetVideoReceiveConfig()
1741 config.local_ssrc = sender_config.ssrcs(0); in GetVideoSendConfig()
1776 config.local_ssrc = receiver_config.local_ssrc(); in GetAudioReceiveConfig()
1795 config.local_ssrc = sender_config.ssrc(); in GetAudioSendConfig()
1815 res.ssrc = playout_event.local_ssrc(); in GetAudioPlayout()
[all …]
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| D | rtc_event_log.proto | 129 // the receive stream, while local_ssrc identifies the send stream, if any.
131 optional uint32 local_ssrc = 2; field
170 optional uint32 local_ssrc = 2; field
263 optional uint32 local_ssrc = 2; field
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| /external/webrtc/audio/voip/ |
| D | voip_core.cc | 121 absl::optional<uint32_t> local_ssrc) { in CreateChannel() argument
125 if (!local_ssrc) { in CreateChannel()
127 local_ssrc = random.Rand<uint32_t>(); in CreateChannel()
132 transport, local_ssrc.value(), task_queue_factory_.get(), in CreateChannel()
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| D | audio_channel.cc | 33 uint32_t local_ssrc, in AudioChannel() argument
52 rtp_config.local_media_ssrc = local_ssrc; in AudioChannel()
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| D | audio_channel.h | 35 uint32_t local_ssrc,
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| D | voip_core.h | 70 absl::optional<uint32_t> local_ssrc) override;
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| /external/webrtc/call/ |
| D | flexfec_receive_stream_impl.cc | 48 ss << ", local_ssrc: " << local_ssrc; in ToString()
133 configuration.local_media_ssrc = config.local_ssrc; in CreateRtpRtcpModule()
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| D | flexfec_receive_stream.h | 63 uint32_t local_ssrc = 0; member
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| D | call.cc | 116 rtclog_config->local_ssrc = config.rtp.local_ssrc; in CreateRtcLogStreamConfig()
134 rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index]; in CreateRtcLogStreamConfig()
151 rtclog_config->local_ssrc = config.rtp.local_ssrc; in CreateRtcLogStreamConfig()
787 if (stream->config().rtp.local_ssrc == config.rtp.ssrc) { in CreateAudioSendStream()
812 if (stream->config().rtp.local_ssrc == ssrc) { in DestroyAudioSendStream()
838 auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc); in CreateAudioReceiveStream()
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| D | audio_receive_stream.h | 112 uint32_t local_ssrc = 0; member
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| D | audio_send_stream.h | 43 uint32_t local_ssrc = 0; member
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| D | bitrate_estimator_tests.cc | 142 receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc; in SetUp()
202 test_->receive_config_.rtp.local_ssrc++;
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| /external/webrtc/audio/ |
| D | audio_receive_stream.cc | 38 ss << ", local_ssrc: " << local_ssrc; in ToString()
81 event_log, config.rtp.local_ssrc, config.rtp.remote_ssrc, in CreateChannelReceive()
398 RTC_DCHECK_EQ(old_config.rtp.local_ssrc, new_config.rtp.local_ssrc); in ConfigureStream()
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| D | channel_receive.cc | 91 uint32_t local_ssrc,
451 uint32_t local_ssrc, in ChannelReceive() argument
505 configuration.local_media_ssrc = local_ssrc; in ChannelReceive()
979 uint32_t local_ssrc, in CreateChannelReceive() argument
992 rtcp_send_transport, rtc_event_log, local_ssrc, remote_ssrc, in CreateChannelReceive()
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| /external/webrtc/api/voip/ |
| D | voip_base.h | 52 absl::optional<uint32_t> local_ssrc) = 0;
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| /external/webrtc/test/fuzzers/ |
| D | vp9_replay_fuzzer.cc | 29 vp9_config.rtp.local_ssrc = 7731; in FuzzOneInput()
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| D | vp8_replay_fuzzer.cc | 29 vp8_config.rtp.local_ssrc = 7731; in FuzzOneInput()
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| /external/webrtc/test/scenario/ |
| D | video_stream.cc | 328 uint32_t local_ssrc, in CreateVideoReceiveStreamConfig() argument
333 recv.rtp.local_ssrc = local_ssrc; in CreateVideoReceiveStreamConfig()
573 flexfec.local_ssrc = recv_config.rtp.local_ssrc; in ReceiveVideoStream()
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| /external/webrtc/audio/test/ |
| D | audio_stats_test.cc | 101 EXPECT_EQ(send_stats.local_ssrc, recv_stats.remote_ssrc); in OnStreamsStopped()
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| /external/webrtc/media/engine/ |
| D | webrtc_video_engine.cc | 1435 config->rtp.local_ssrc = rtcp_receiver_report_ssrc_; in ConfigureReceiverRtp()
1441 if (config->rtp.remote_ssrc == config->rtp.local_ssrc) { in ConfigureReceiverRtp()
1442 if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) { in ConfigureReceiverRtp()
1443 config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc; in ConfigureReceiverRtp()
1445 config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1; in ConfigureReceiverRtp()
1470 flexfec_config->local_ssrc = config->rtp.local_ssrc; in ConfigureReceiverRtp()
2810 uint32_t local_ssrc) { in SetLocalSsrc() argument
2815 if (local_ssrc == config_.rtp.local_ssrc) { in SetLocalSsrc()
2818 << local_ssrc; in SetLocalSsrc()
2822 config_.rtp.local_ssrc = local_ssrc; in SetLocalSsrc()
[all …]
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| /external/webrtc/video/ |
| D | rtp_video_stream_receiver2.cc | 86 uint32_t local_ssrc) { in CreateRtpRtcpModule() argument
97 configuration.local_media_ssrc = local_ssrc; in CreateRtpRtcpModule()
237 config_.rtp.local_ssrc)), in RtpVideoStreamReceiver2()
260 RTC_DCHECK(config_.rtp.local_ssrc != 0); in RtpVideoStreamReceiver2()
261 RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc); in RtpVideoStreamReceiver2()
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