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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/format_macros.h" | 22 #include "webrtc/base/format_macros.h" |
23 #include "webrtc/base/optional.h" | 23 #include "webrtc/base/optional.h" |
24 #include "webrtc/base/platform_file.h" | 24 #include "webrtc/base/platform_file.h" |
25 #include "webrtc/base/timeutils.h" | 25 #include "webrtc/base/timeutils.h" |
26 #include "webrtc/call.h" | 26 #include "webrtc/call.h" |
27 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" | 27 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
28 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 28 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
29 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 29 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
30 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 30 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
31 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 31 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| 32 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" |
| 33 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
| 34 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
32 #include "webrtc/system_wrappers/include/cpu_info.h" | 35 #include "webrtc/system_wrappers/include/cpu_info.h" |
33 #include "webrtc/test/gtest.h" | 36 #include "webrtc/test/gtest.h" |
34 #include "webrtc/test/layer_filtering_transport.h" | 37 #include "webrtc/test/layer_filtering_transport.h" |
35 #include "webrtc/test/run_loop.h" | 38 #include "webrtc/test/run_loop.h" |
36 #include "webrtc/test/statistics.h" | 39 #include "webrtc/test/statistics.h" |
37 #include "webrtc/test/testsupport/fileutils.h" | 40 #include "webrtc/test/testsupport/fileutils.h" |
38 #include "webrtc/test/vcm_capturer.h" | 41 #include "webrtc/test/vcm_capturer.h" |
39 #include "webrtc/test/video_renderer.h" | 42 #include "webrtc/test/video_renderer.h" |
40 #include "webrtc/voice_engine/include/voe_base.h" | 43 #include "webrtc/voice_engine/include/voe_base.h" |
41 | 44 |
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999 void VideoQualityTest::SetupVideo(Transport* send_transport, | 1002 void VideoQualityTest::SetupVideo(Transport* send_transport, |
1000 Transport* recv_transport) { | 1003 Transport* recv_transport) { |
1001 if (params_.logs) | 1004 if (params_.logs) |
1002 trace_to_stderr_.reset(new test::TraceToStderr); | 1005 trace_to_stderr_.reset(new test::TraceToStderr); |
1003 | 1006 |
1004 size_t num_streams = params_.ss.streams.size(); | 1007 size_t num_streams = params_.ss.streams.size(); |
1005 CreateSendConfig(num_streams, 0, 0, send_transport); | 1008 CreateSendConfig(num_streams, 0, 0, send_transport); |
1006 | 1009 |
1007 int payload_type; | 1010 int payload_type; |
1008 if (params_.video.codec == "H264") { | 1011 if (params_.video.codec == "H264") { |
1009 video_encoder_.reset(VideoEncoder::Create(VideoEncoder::kH264)); | 1012 video_encoder_.reset(H264Encoder::Create()); |
1010 payload_type = kPayloadTypeH264; | 1013 payload_type = kPayloadTypeH264; |
1011 } else if (params_.video.codec == "VP8") { | 1014 } else if (params_.video.codec == "VP8") { |
1012 video_encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp8)); | 1015 video_encoder_.reset(VP8Encoder::Create()); |
1013 payload_type = kPayloadTypeVP8; | 1016 payload_type = kPayloadTypeVP8; |
1014 } else if (params_.video.codec == "VP9") { | 1017 } else if (params_.video.codec == "VP9") { |
1015 video_encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp9)); | 1018 video_encoder_.reset(VP9Encoder::Create()); |
1016 payload_type = kPayloadTypeVP9; | 1019 payload_type = kPayloadTypeVP9; |
1017 } else { | 1020 } else { |
1018 RTC_NOTREACHED() << "Codec not supported!"; | 1021 RTC_NOTREACHED() << "Codec not supported!"; |
1019 return; | 1022 return; |
1020 } | 1023 } |
1021 video_send_config_.encoder_settings.encoder = video_encoder_.get(); | 1024 video_send_config_.encoder_settings.encoder = video_encoder_.get(); |
1022 video_send_config_.encoder_settings.payload_name = params_.video.codec; | 1025 video_send_config_.encoder_settings.payload_name = params_.video.codec; |
1023 video_send_config_.encoder_settings.payload_type = payload_type; | 1026 video_send_config_.encoder_settings.payload_type = payload_type; |
1024 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | 1027 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
1025 video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; | 1028 video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; |
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1482 std::ostringstream str; | 1485 std::ostringstream str; |
1483 str << receive_logs_++; | 1486 str << receive_logs_++; |
1484 std::string path = | 1487 std::string path = |
1485 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; | 1488 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; |
1486 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), | 1489 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), |
1487 10000000); | 1490 10000000); |
1488 } | 1491 } |
1489 } | 1492 } |
1490 | 1493 |
1491 } // namespace webrtc | 1494 } // namespace webrtc |
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