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Side by Side Diff: webrtc/tools/event_log_visualizer/analyzer.cc

Issue 2851303007: Replace AudioReceiveStream::Config with rtclog::StreamConfig. (Closed)
Patch Set: Rebased. Created 3 years, 7 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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350 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions); 350 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
351 video_ssrcs_.insert(stream); 351 video_ssrcs_.insert(stream);
352 StreamId rtx_stream(config.rtx_ssrc, kOutgoingPacket); 352 StreamId rtx_stream(config.rtx_ssrc, kOutgoingPacket);
353 extension_maps[rtx_stream] = 353 extension_maps[rtx_stream] =
354 RtpHeaderExtensionMap(config.rtp_extensions); 354 RtpHeaderExtensionMap(config.rtp_extensions);
355 video_ssrcs_.insert(rtx_stream); 355 video_ssrcs_.insert(rtx_stream);
356 rtx_ssrcs_.insert(rtx_stream); 356 rtx_ssrcs_.insert(rtx_stream);
357 break; 357 break;
358 } 358 }
359 case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: { 359 case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
360 AudioReceiveStream::Config config; 360 rtclog::StreamConfig config;
361 parsed_log_.GetAudioReceiveConfig(i, &config); 361 parsed_log_.GetAudioReceiveConfig(i, &config);
362 StreamId stream(config.rtp.remote_ssrc, kIncomingPacket); 362 StreamId stream(config.remote_ssrc, kIncomingPacket);
363 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions); 363 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
364 audio_ssrcs_.insert(stream); 364 audio_ssrcs_.insert(stream);
365 break; 365 break;
366 } 366 }
367 case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: { 367 case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
368 AudioSendStream::Config config(nullptr); 368 AudioSendStream::Config config(nullptr);
369 parsed_log_.GetAudioSendConfig(i, &config); 369 parsed_log_.GetAudioSendConfig(i, &config);
370 StreamId stream(config.rtp.ssrc, kOutgoingPacket); 370 StreamId stream(config.rtp.ssrc, kOutgoingPacket);
371 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions); 371 extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
372 audio_ssrcs_.insert(stream); 372 audio_ssrcs_.insert(stream);
373 break; 373 break;
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1395 }, 1395 },
1396 audio_network_adaptation_events_, begin_time_, &time_series); 1396 audio_network_adaptation_events_, begin_time_, &time_series);
1397 plot->AppendTimeSeries(std::move(time_series)); 1397 plot->AppendTimeSeries(std::move(time_series));
1398 plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin); 1398 plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
1399 plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))", 1399 plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
1400 kBottomMargin, kTopMargin); 1400 kBottomMargin, kTopMargin);
1401 plot->SetTitle("Reported audio encoder number of channels"); 1401 plot->SetTitle("Reported audio encoder number of channels");
1402 } 1402 }
1403 } // namespace plotting 1403 } // namespace plotting
1404 } // namespace webrtc 1404 } // namespace webrtc
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