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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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| 29 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" | 29 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" |
| 30 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | 30 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
| 31 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h" | 31 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h" |
| 32 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 32 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
| 33 #include "webrtc/rtc_base/checks.h" | 33 #include "webrtc/rtc_base/checks.h" |
| 34 #include "webrtc/rtc_base/cpu_time.h" | 34 #include "webrtc/rtc_base/cpu_time.h" |
| 35 #include "webrtc/rtc_base/event.h" | 35 #include "webrtc/rtc_base/event.h" |
| 36 #include "webrtc/rtc_base/format_macros.h" | 36 #include "webrtc/rtc_base/format_macros.h" |
| 37 #include "webrtc/rtc_base/memory_usage.h" | 37 #include "webrtc/rtc_base/memory_usage.h" |
| 38 #include "webrtc/rtc_base/optional.h" | 38 #include "webrtc/rtc_base/optional.h" |
| 39 #include "webrtc/rtc_base/pathutils.h" | |
| 39 #include "webrtc/rtc_base/platform_file.h" | 40 #include "webrtc/rtc_base/platform_file.h" |
| 40 #include "webrtc/rtc_base/timeutils.h" | 41 #include "webrtc/rtc_base/timeutils.h" |
| 41 #include "webrtc/system_wrappers/include/cpu_info.h" | 42 #include "webrtc/system_wrappers/include/cpu_info.h" |
| 42 #include "webrtc/system_wrappers/include/field_trial.h" | 43 #include "webrtc/system_wrappers/include/field_trial.h" |
| 43 #include "webrtc/test/gtest.h" | 44 #include "webrtc/test/gtest.h" |
| 44 #include "webrtc/test/layer_filtering_transport.h" | 45 #include "webrtc/test/layer_filtering_transport.h" |
| 45 #include "webrtc/test/run_loop.h" | 46 #include "webrtc/test/run_loop.h" |
| 46 #include "webrtc/test/statistics.h" | 47 #include "webrtc/test/statistics.h" |
| 47 #include "webrtc/test/testsupport/fileutils.h" | 48 #include "webrtc/test/testsupport/fileutils.h" |
| 48 #include "webrtc/test/testsupport/frame_writer.h" | 49 #include "webrtc/test/testsupport/frame_writer.h" |
| 50 #include "webrtc/test/testsupport/test_output.h" | |
| 49 #include "webrtc/test/vcm_capturer.h" | 51 #include "webrtc/test/vcm_capturer.h" |
| 50 #include "webrtc/test/video_renderer.h" | 52 #include "webrtc/test/video_renderer.h" |
| 51 #include "webrtc/voice_engine/include/voe_base.h" | 53 #include "webrtc/voice_engine/include/voe_base.h" |
| 52 | 54 |
| 53 #include "webrtc/test/rtp_file_writer.h" | 55 #include "webrtc/test/rtp_file_writer.h" |
| 54 | 56 |
| 55 namespace { | 57 namespace { |
| 56 | 58 |
| 57 constexpr int kSendStatsPollingIntervalMs = 1000; | 59 constexpr int kSendStatsPollingIntervalMs = 1000; |
| 58 | 60 |
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| 817 dropped_frames_); | 819 dropped_frames_); |
| 818 printf("RESULT cpu_usage: %s = %lf %%\n", test_label_.c_str(), | 820 printf("RESULT cpu_usage: %s = %lf %%\n", test_label_.c_str(), |
| 819 GetCpuUsagePercent()); | 821 GetCpuUsagePercent()); |
| 820 | 822 |
| 821 #if defined(WEBRTC_WIN) | 823 #if defined(WEBRTC_WIN) |
| 822 // On Linux and Mac in Resident Set some unused pages may be counted. | 824 // On Linux and Mac in Resident Set some unused pages may be counted. |
| 823 // Therefore this metric will depend on order in which tests are run and | 825 // Therefore this metric will depend on order in which tests are run and |
| 824 // will be flaky. | 826 // will be flaky. |
| 825 PrintResult("memory_usage", memory_usage_, " bytes"); | 827 PrintResult("memory_usage", memory_usage_, " bytes"); |
| 826 #endif | 828 #endif |
| 827 // TODO(ilnik): enable frame writing for android, once jpeg frame writer | |
| 828 // is implemented. | |
| 829 | 829 |
| 830 #if !defined(WEBRTC_ANDROID) | 830 // LibJpeg is not available on iOS. |
| 831 #if !defined(WEBRTC_IOS) | |
| 832 // Saving only the worst frame for manual analysis. Intention here is to | |
| 833 // only detect video corruptions and not to track picture quality. Thus, | |
| 834 // jpeg is used here. | |
| 831 if (worst_frame_) { | 835 if (worst_frame_) { |
| 832 test::Y4mFrameWriterImpl frame_writer(test_label_ + ".y4m", | 836 std::string output_dir; |
| 833 worst_frame_->frame.width(), | 837 if (!test::GetTestOutputDir(&output_dir)) { |
| 834 worst_frame_->frame.height(), 1); | 838 output_dir = ""; |
|
sprang_webrtc
2017/08/17 15:27:50
Should not write to file instead, if flag is not s
ilnik
2017/08/18 08:42:05
Currently it always outputs the worst frame. This
| |
| 835 bool res = frame_writer.Init(); | 839 } |
| 836 RTC_DCHECK(res); | 840 std::string output_path = |
| 837 size_t length = | 841 rtc::Pathname(output_dir, test_label_ + ".jpg").pathname(); |
| 838 CalcBufferSize(VideoType::kI420, worst_frame_->frame.width(), | 842 fprintf(stderr, "Saving worst frame to %s\n", output_path.c_str()); |
|
sprang_webrtc
2017/08/17 15:27:50
use LOG macro instead
ilnik
2017/08/18 08:42:05
Done.
| |
| 839 worst_frame_->frame.height()); | 843 test::JpegFrameWriter frame_writer(output_path); |
| 840 rtc::Buffer extracted_buffer(length); | 844 RTC_CHECK(frame_writer.WriteFrame(worst_frame_->frame, |
| 841 size_t extracted_length = | 845 100 /*best quality*/)); |
| 842 ExtractBuffer(worst_frame_->frame.video_frame_buffer()->ToI420(), | |
| 843 length, extracted_buffer.data()); | |
| 844 RTC_DCHECK_EQ(extracted_length, frame_writer.FrameLength()); | |
| 845 res = frame_writer.WriteFrame(extracted_buffer.data()); | |
| 846 RTC_DCHECK(res); | |
| 847 frame_writer.Close(); | |
| 848 } | 846 } |
| 849 #endif | 847 #endif |
| 850 | 848 |
| 851 // Disable quality check for quick test, as quality checks may fail | 849 // Disable quality check for quick test, as quality checks may fail |
| 852 // because too few samples were collected. | 850 // because too few samples were collected. |
| 853 if (!is_quick_test_enabled_) { | 851 if (!is_quick_test_enabled_) { |
| 854 EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_); | 852 EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_); |
| 855 EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_); | 853 EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_); |
| 856 } | 854 } |
| 857 } | 855 } |
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| 2119 if (!params_.logging.encoded_frame_base_path.empty()) { | 2117 if (!params_.logging.encoded_frame_base_path.empty()) { |
| 2120 std::ostringstream str; | 2118 std::ostringstream str; |
| 2121 str << receive_logs_++; | 2119 str << receive_logs_++; |
| 2122 std::string path = | 2120 std::string path = |
| 2123 params_.logging.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; | 2121 params_.logging.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; |
| 2124 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), | 2122 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), |
| 2125 100000000); | 2123 100000000); |
| 2126 } | 2124 } |
| 2127 } | 2125 } |
| 2128 } // namespace webrtc | 2126 } // namespace webrtc |
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