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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 #include "webrtc/video/video_quality_test.h" | 10 #include "webrtc/video/video_quality_test.h" |
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22 #include "webrtc/base/event.h" | 22 #include "webrtc/base/event.h" |
23 #include "webrtc/base/format_macros.h" | 23 #include "webrtc/base/format_macros.h" |
24 #include "webrtc/base/optional.h" | 24 #include "webrtc/base/optional.h" |
25 #include "webrtc/base/platform_file.h" | 25 #include "webrtc/base/platform_file.h" |
26 #include "webrtc/base/timeutils.h" | 26 #include "webrtc/base/timeutils.h" |
27 #include "webrtc/call.h" | 27 #include "webrtc/call.h" |
28 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" | 28 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
30 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 30 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
31 #include "webrtc/system_wrappers/include/cpu_info.h" | 31 #include "webrtc/system_wrappers/include/cpu_info.h" |
32 #include "webrtc/test/encoder_settings.h" | |
33 #include "webrtc/test/layer_filtering_transport.h" | 32 #include "webrtc/test/layer_filtering_transport.h" |
34 #include "webrtc/test/run_loop.h" | 33 #include "webrtc/test/run_loop.h" |
35 #include "webrtc/test/statistics.h" | 34 #include "webrtc/test/statistics.h" |
36 #include "webrtc/test/testsupport/fileutils.h" | 35 #include "webrtc/test/testsupport/fileutils.h" |
37 #include "webrtc/test/vcm_capturer.h" | 36 #include "webrtc/test/vcm_capturer.h" |
38 #include "webrtc/test/video_renderer.h" | 37 #include "webrtc/test/video_renderer.h" |
39 #include "webrtc/voice_engine/include/voe_base.h" | 38 #include "webrtc/voice_engine/include/voe_base.h" |
40 #include "webrtc/voice_engine/include/voe_codec.h" | 39 #include "webrtc/voice_engine/include/voe_codec.h" |
41 | 40 |
42 namespace { | 41 namespace { |
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1008 video_send_config_.rtp.extensions.push_back( | 1007 video_send_config_.rtp.extensions.push_back( |
1009 RtpExtension(RtpExtension::kTransportSequenceNumberUri, | 1008 RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
1010 test::kTransportSequenceNumberExtensionId)); | 1009 test::kTransportSequenceNumberExtensionId)); |
1011 } else { | 1010 } else { |
1012 video_send_config_.rtp.extensions.push_back(RtpExtension( | 1011 video_send_config_.rtp.extensions.push_back(RtpExtension( |
1013 RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId)); | 1012 RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId)); |
1014 } | 1013 } |
1015 | 1014 |
1016 video_encoder_config_.min_transmit_bitrate_bps = | 1015 video_encoder_config_.min_transmit_bitrate_bps = |
1017 params_.common.min_transmit_bps; | 1016 params_.common.min_transmit_bps; |
1018 test::FillEncoderConfiguration(params_.ss.streams.size(), | 1017 video_encoder_config_.streams = params_.ss.streams; |
1019 &video_encoder_config_); | |
1020 video_encoder_config_.spatial_layers = params_.ss.spatial_layers; | 1018 video_encoder_config_.spatial_layers = params_.ss.spatial_layers; |
1021 | 1019 |
1022 CreateMatchingReceiveConfigs(recv_transport); | 1020 CreateMatchingReceiveConfigs(recv_transport); |
1023 | 1021 |
1024 for (size_t i = 0; i < num_streams; ++i) { | 1022 for (size_t i = 0; i < num_streams; ++i) { |
1025 video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | 1023 video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
1026 video_receive_configs_[i].rtp.rtx[payload_type].ssrc = kSendRtxSsrcs[i]; | 1024 video_receive_configs_[i].rtp.rtx[payload_type].ssrc = kSendRtxSsrcs[i]; |
1027 video_receive_configs_[i].rtp.rtx[payload_type].payload_type = | 1025 video_receive_configs_[i].rtp.rtx[payload_type].payload_type = |
1028 kSendRtxPayloadType; | 1026 kSendRtxPayloadType; |
1029 video_receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe; | 1027 video_receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe; |
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1392 std::ostringstream str; | 1390 std::ostringstream str; |
1393 str << receive_logs_++; | 1391 str << receive_logs_++; |
1394 std::string path = | 1392 std::string path = |
1395 params_.common.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; | 1393 params_.common.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; |
1396 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), | 1394 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), |
1397 10000000); | 1395 10000000); |
1398 } | 1396 } |
1399 } | 1397 } |
1400 | 1398 |
1401 } // namespace webrtc | 1399 } // namespace webrtc |
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