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Issue 2826263004: Move responsibility for RTP header extensions on video receive. (Closed)
Patch Set: Created 3 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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324 uint64_t timestamp = parsed_log_.GetTimestamp(i); 324 uint64_t timestamp = parsed_log_.GetTimestamp(i);
325 first_timestamp = std::min(first_timestamp, timestamp); 325 first_timestamp = std::min(first_timestamp, timestamp);
326 last_timestamp = std::max(last_timestamp, timestamp); 326 last_timestamp = std::max(last_timestamp, timestamp);
327 } 327 }
328 328
329 switch (parsed_log_.GetEventType(i)) { 329 switch (parsed_log_.GetEventType(i)) {
330 case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: { 330 case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
331 VideoReceiveStream::Config config(nullptr); 331 VideoReceiveStream::Config config(nullptr);
332 parsed_log_.GetVideoReceiveConfig(i, &config); 332 parsed_log_.GetVideoReceiveConfig(i, &config);
333 StreamId stream(config.rtp.remote_ssrc, kIncomingPacket); 333 StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
334 #if 0
334 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions); 335 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
336 #endif
335 video_ssrcs_.insert(stream); 337 video_ssrcs_.insert(stream);
336 StreamId rtx_stream(config.rtp.rtx_ssrc, kIncomingPacket); 338 StreamId rtx_stream(config.rtp.rtx_ssrc, kIncomingPacket);
339 #if 0
337 extension_maps[rtx_stream] = 340 extension_maps[rtx_stream] =
338 RtpHeaderExtensionMap(config.rtp.extensions); 341 RtpHeaderExtensionMap(config.rtp.extensions);
342 #endif
339 video_ssrcs_.insert(rtx_stream); 343 video_ssrcs_.insert(rtx_stream);
340 rtx_ssrcs_.insert(rtx_stream); 344 rtx_ssrcs_.insert(rtx_stream);
341 break; 345 break;
342 } 346 }
343 case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: { 347 case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
344 VideoSendStream::Config config(nullptr); 348 VideoSendStream::Config config(nullptr);
345 parsed_log_.GetVideoSendConfig(i, &config); 349 parsed_log_.GetVideoSendConfig(i, &config);
346 for (auto ssrc : config.rtp.ssrcs) { 350 for (auto ssrc : config.rtp.ssrcs) {
347 StreamId stream(ssrc, kOutgoingPacket); 351 StreamId stream(ssrc, kOutgoingPacket);
348 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions); 352 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
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1388 }, 1392 },
1389 audio_network_adaptation_events_, begin_time_, &time_series); 1393 audio_network_adaptation_events_, begin_time_, &time_series);
1390 plot->AppendTimeSeries(std::move(time_series)); 1394 plot->AppendTimeSeries(std::move(time_series));
1391 plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin); 1395 plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
1392 plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))", 1396 plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
1393 kBottomMargin, kTopMargin); 1397 kBottomMargin, kTopMargin);
1394 plot->SetTitle("Reported audio encoder number of channels"); 1398 plot->SetTitle("Reported audio encoder number of channels");
1395 } 1399 }
1396 } // namespace plotting 1400 } // namespace plotting
1397 } // namespace webrtc 1401 } // namespace webrtc
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