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Issue 1374233002: Collecting encode_time_ms for each frame (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressing comments Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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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155 config_.rtp.fec.ulpfec_payload_type); 155 config_.rtp.fec.ulpfec_payload_type);
156 vie_encoder_->UpdateProtectionMethod(enable_protection_nack, 156 vie_encoder_->UpdateProtectionMethod(enable_protection_nack,
157 enable_protection_fec); 157 enable_protection_fec);
158 158
159 ConfigureSsrcs(); 159 ConfigureSsrcs();
160 160
161 vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str()); 161 vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str());
162 162
163 input_.reset(new internal::VideoCaptureInput( 163 input_.reset(new internal::VideoCaptureInput(
164 module_process_thread_, vie_encoder_, config_.local_renderer, 164 module_process_thread_, vie_encoder_, config_.local_renderer,
165 &stats_proxy_, this)); 165 &stats_proxy_, this, config_.encoding_time_observer));
166 166
167 // 28 to match packet overhead in ModuleRtpRtcpImpl. 167 // 28 to match packet overhead in ModuleRtpRtcpImpl.
168 RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28)); 168 RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28));
169 vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28)); 169 vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28));
170 170
171 RTC_DCHECK(config.encoder_settings.encoder != nullptr); 171 RTC_DCHECK(config.encoder_settings.encoder != nullptr);
172 RTC_DCHECK_GE(config.encoder_settings.payload_type, 0); 172 RTC_DCHECK_GE(config.encoder_settings.payload_type, 0);
173 RTC_DCHECK_LE(config.encoder_settings.payload_type, 127); 173 RTC_DCHECK_LE(config.encoder_settings.payload_type, 127);
174 RTC_CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder( 174 RTC_CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder(
175 config.encoder_settings.encoder, 175 config.encoder_settings.encoder,
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502 vie_channel_->IsSendingFecEnabled()); 502 vie_channel_->IsSendingFecEnabled());
503 503
504 // Restart the media flow 504 // Restart the media flow
505 vie_encoder_->Restart(); 505 vie_encoder_->Restart();
506 506
507 return true; 507 return true;
508 } 508 }
509 509
510 } // namespace internal 510 } // namespace internal
511 } // namespace webrtc 511 } // namespace webrtc
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