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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/jitter/delay_variation_calculator.h
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/jitter/delay_variation_calculator.h')
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+/*
+ * Copyright (c) 2023 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.
+ */
+
+#ifndef TEST_JITTER_DELAY_VARIATION_CALCULATOR_H_
+#define TEST_JITTER_DELAY_VARIATION_CALCULATOR_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_frame_type.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+namespace test {
+
+// Helper class for calculating different delay variation statistics for "RTP
+// frame arrival events". One use case is gathering statistics from
+// RtcEventLogs. Another use case is online logging of data in test calls.
+class DelayVariationCalculator {
+ public:
+ struct TimeSeries {
+ // Time series of RTP timestamps `t(n)` for each frame `n`.
+ SamplesStatsCounter rtp_timestamps;
+ // Time series of local arrival timestamps `r(n)` for each frame.
+ SamplesStatsCounter arrival_times_ms;
+ // Time series of sizes `s(n)` for each frame.
+ SamplesStatsCounter sizes_bytes;
+ // Time series of `d_t(n) = t(n) - t(n-1)` for each frame.
+ SamplesStatsCounter inter_departure_times_ms;
+ // Time series of `d_r(n) = r(n) - r(n-1)` for each frame.
+ SamplesStatsCounter inter_arrival_times_ms;
+ // Time series of `d_r(n) - d_t(n) = (r(n) - r(n-1)) - (t(n) - t(n-1))`
+ // for each frame.
+ SamplesStatsCounter inter_delay_variations_ms;
+ // Time series of `s(n) - s(n-1)`, for each frame.
+ SamplesStatsCounter inter_size_variations_bytes;
+ };
+
+ DelayVariationCalculator() = default;
+ ~DelayVariationCalculator() = default;
+
+ void Insert(uint32_t rtp_timestamp,
+ Timestamp arrival_time,
+ DataSize size,
+ absl::optional<int> spatial_layer = absl::nullopt,
+ absl::optional<int> temporal_layer = absl::nullopt,
+ absl::optional<VideoFrameType> frame_type = absl::nullopt);
+
+ const TimeSeries& time_series() const { return time_series_; }
+
+ private:
+ struct Frame {
+ uint32_t rtp_timestamp;
+ int64_t unwrapped_rtp_timestamp;
+ Timestamp arrival_time;
+ DataSize size;
+ absl::optional<int> spatial_layer;
+ absl::optional<int> temporal_layer;
+ absl::optional<VideoFrameType> frame_type;
+ };
+ using MetadataT = std::map<std::string, std::string>;
+
+ void InsertFirstFrame(const Frame& frame,
+ Timestamp sample_time,
+ MetadataT sample_metadata);
+ void InsertFrame(const Frame& frame,
+ Timestamp sample_time,
+ MetadataT sample_metadata);
+
+ MetadataT BuildMetadata(const Frame& frame);
+
+ RtpTimestampUnwrapper unwrapper_;
+ absl::optional<Frame> prev_frame_ = absl::nullopt;
+ TimeSeries time_series_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // TEST_JITTER_DELAY_VARIATION_CALCULATOR_H_