blob: 6400f82ad738fdd862f4ae900bfe2db3447556f0 [file] [log] [blame]
* 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.
#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 {
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_; }
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