| /external/webrtc/webrtc/ |
| H A D | video_receive_stream.h | 129 RtxMap rtx; member in struct:webrtc::VideoReceiveStream::Config::Rtp
131 // If set to true, the RTX payload type mapping supplied in |rtx| will be
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| H A D | video_send_stream.h | 129 } rtx; member in struct:webrtc::VideoSendStream::Config::Rtp
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| /external/webrtc/webrtc/call/ |
| H A D | rampup_tests.cc | 35 bool rtx,
42 rtx_(rtx),
144 send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
145 send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
170 recv_config.rtp.rtx[send_config->encoder_settings.payload_type].ssrc =
172 recv_config.rtp.rtx[send_config->encoder_settings.payload_type]
173 .payload_type = send_config->rtp.rtx.payload_type;
295 ReportResult("ramp-up-rtx-total-packets-sent", rtx_total_packets_sent,
297 ReportResult("ramp-up-rtx-total-sent", rtx_total_sent, "bytes");
298 ReportResult("ramp-up-rtx
31 RampUpTester(size_t num_video_streams, size_t num_audio_streams, unsigned int start_bitrate_bps, const std::string& extension_type, bool rtx, bool red) argument
316 RampUpDownUpTester(size_t num_video_streams, size_t num_audio_streams, unsigned int start_bitrate_bps, const std::string& extension_type, bool rtx, bool red) argument
[all...] |
| H A D | rtc_event_log_unittest.cc | 145 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
150 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
152 const VideoReceiveStream::Config::Rtp::Rtx& rtx = local
153 config.rtp.rtx.at(rtx_map.payload_type());
156 EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
157 EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
206 ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
209 EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
213 EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
385 config->rtp.rtx
[all...] |
| /external/webrtc/talk/session/media/ |
| H A D | mediasession_unittest.cc | 215 VideoCodec rtx; local
216 ASSERT_FALSE(cricket::FindCodecById(*codecs, rtx_codec.id, &rtx));
1576 // This creates rtx for H264 with the payload type |f1_| uses.
1581 // This creates rtx for H264 with the payload type |f2_| uses.
1623 // This creates rtx for H264 with the payload type |f1_| uses.
1668 VideoCodec rtx = updated_vcd->codecs()[1]; local
1670 rtx.params[cricket::kCodecParamAssociatedPayloadType]);
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| /external/webrtc/webrtc/video/ |
| H A D | vie_channel.cc | 237 StreamDataCounters rtx; local
238 GetSendStreamDataCounters(&rtp, &rtx);
240 rtp_rtx.Add(rtx);
263 static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
302 StreamDataCounters rtx; local
303 GetReceiveStreamDataCounters(&rtp, &rtx);
305 rtp_rtx.Add(rtx);
327 static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
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| /external/webrtc/talk/app/webrtc/ |
| H A D | webrtcsdp_unittest.cc | 1306 cricket::VideoCodec rtx = vcd->codecs()[1]; local
1307 EXPECT_EQ("RTX", rtx.name);
1308 EXPECT_EQ(95, rtx.id);
1309 VerifyCodecParameter(rtx.params, "apt", vp8.id);
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| /external/webrtc/talk/media/webrtc/ |
| H A D | webrtcvideoengine2.cc | 334 // TODO(andresp): Add rtx codec for vp9 and verify it works.
1140 webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx = local
1141 config->rtp.rtx[recv_codecs_[i].codec.id];
1142 rtx.ssrc = rtx_ssrc;
1143 rtx.payload_type = recv_codecs_[i].rtx_payload_type;
1626 ¶meters_.config.rtp.rtx.ssrcs);
1907 if (!parameters_.config.rtp.rtx.ssrcs.empty()) {
1911 parameters_.config.rtp.rtx.ssrcs.clear();
1913 parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
2215 if (!config.rtp.rtx
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_sender.cc | 385 int rtx; local
388 rtx = rtx_;
395 - ((rtx) ? 2 : 0); // RTX overhead.
738 int rtx = kRtxOff; local
741 rtx = rtx_;
744 (rtx & kRtxRetransmitted) > 0, true)) {
894 int rtx; local
897 rtx = rtx_;
902 retransmission && (rtx & kRtxRetransmitted) > 0,
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