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-rw-r--r--third_party/libwebrtc/test/scenario/performance_stats.cc47
1 files changed, 47 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/scenario/performance_stats.cc b/third_party/libwebrtc/test/scenario/performance_stats.cc
new file mode 100644
index 0000000000..e12be8a003
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/performance_stats.cc
@@ -0,0 +1,47 @@
+/*
+ * Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/performance_stats.h"
+
+#include <algorithm>
+
+namespace webrtc {
+namespace test {
+void VideoFramesStats::AddFrameInfo(const VideoFrameBuffer& frame,
+ Timestamp at_time) {
+ ++count;
+ RTC_DCHECK(at_time.IsFinite());
+ pixels.AddSample(frame.width() * frame.height());
+ resolution.AddSample(std::max(frame.width(), frame.height()));
+ frames.AddEvent(at_time);
+}
+
+void VideoFramesStats::AddStats(const VideoFramesStats& other) {
+ count += other.count;
+ pixels.AddSamples(other.pixels);
+ resolution.AddSamples(other.resolution);
+ frames.AddEvents(other.frames);
+}
+
+void VideoQualityStats::AddStats(const VideoQualityStats& other) {
+ capture.AddStats(other.capture);
+ render.AddStats(other.render);
+ lost_count += other.lost_count;
+ freeze_count += other.freeze_count;
+ capture_to_decoded_delay.AddSamples(other.capture_to_decoded_delay);
+ end_to_end_delay.AddSamples(other.end_to_end_delay);
+ psnr.AddSamples(other.psnr);
+ psnr_with_freeze.AddSamples(other.psnr_with_freeze);
+ skipped_between_rendered.AddSamples(other.skipped_between_rendered);
+ freeze_duration.AddSamples(other.freeze_duration);
+ time_between_freezes.AddSamples(other.time_between_freezes);
+}
+
+} // namespace test
+} // namespace webrtc