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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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358 } | 358 } |
359 case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: { | 359 case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: { |
360 rtclog::StreamConfig config; | 360 rtclog::StreamConfig config; |
361 parsed_log_.GetAudioReceiveConfig(i, &config); | 361 parsed_log_.GetAudioReceiveConfig(i, &config); |
362 StreamId stream(config.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 rtclog::StreamConfig config; |
369 parsed_log_.GetAudioSendConfig(i, &config); | 369 parsed_log_.GetAudioSendConfig(i, &config); |
370 StreamId stream(config.rtp.ssrc, kOutgoingPacket); | 370 StreamId stream(config.local_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; |
374 } | 374 } |
375 case ParsedRtcEventLog::RTP_EVENT: { | 375 case ParsedRtcEventLog::RTP_EVENT: { |
376 MediaType media_type; | 376 MediaType media_type; |
377 parsed_log_.GetRtpHeader(i, &direction, &media_type, header, | 377 parsed_log_.GetRtpHeader(i, &direction, &media_type, header, |
378 &header_length, &total_length); | 378 &header_length, &total_length); |
379 // Parse header to get SSRC. | 379 // Parse header to get SSRC. |
380 RtpUtility::RtpHeaderParser rtp_parser(header, header_length); | 380 RtpUtility::RtpHeaderParser rtp_parser(header, header_length); |
381 RTPHeader parsed_header; | 381 RTPHeader parsed_header; |
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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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