mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00

Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression. This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet. Bug: webrtc:14852 Change-Id: I967160069055036f93e595d328c4d5f1ca483be9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39840}
278 lines
8.5 KiB
C++
278 lines
8.5 KiB
C++
/*
|
|
* 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 "modules/video_coding/codecs/test/video_codec_stats_impl.h"
|
|
|
|
#include <algorithm>
|
|
|
|
#include "api/numerics/samples_stats_counter.h"
|
|
#include "api/test/metrics/metrics_logger.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/time_utils.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
namespace {
|
|
using Frame = VideoCodecStats::Frame;
|
|
using Stream = VideoCodecStats::Stream;
|
|
|
|
constexpr Frequency k90kHz = Frequency::Hertz(90000);
|
|
|
|
class LeakyBucket {
|
|
public:
|
|
LeakyBucket() : level_bits_(0) {}
|
|
|
|
// Updates bucket level and returns its current level in bits. Data is remove
|
|
// from bucket with rate equal to target bitrate of previous frame. Bucket
|
|
// level is tracked with floating point precision. Returned value of bucket
|
|
// level is rounded up.
|
|
int Update(const Frame& frame) {
|
|
RTC_CHECK(frame.target_bitrate) << "Bitrate must be specified.";
|
|
|
|
if (prev_frame_) {
|
|
RTC_CHECK_GT(frame.timestamp_rtp, prev_frame_->timestamp_rtp)
|
|
<< "Timestamp must increase.";
|
|
TimeDelta passed =
|
|
(frame.timestamp_rtp - prev_frame_->timestamp_rtp) / k90kHz;
|
|
level_bits_ -=
|
|
prev_frame_->target_bitrate->bps() * passed.us() / 1000000.0;
|
|
level_bits_ = std::max(level_bits_, 0.0);
|
|
}
|
|
|
|
prev_frame_ = frame;
|
|
|
|
level_bits_ += frame.frame_size.bytes() * 8;
|
|
return static_cast<int>(std::ceil(level_bits_));
|
|
}
|
|
|
|
private:
|
|
absl::optional<Frame> prev_frame_;
|
|
double level_bits_;
|
|
};
|
|
|
|
// Merges spatial layer frames into superframes.
|
|
std::vector<Frame> Merge(const std::vector<Frame>& frames) {
|
|
std::vector<Frame> superframes;
|
|
// Map from frame timestamp to index in `superframes` vector.
|
|
std::map<uint32_t, int> index;
|
|
|
|
for (const auto& f : frames) {
|
|
if (index.find(f.timestamp_rtp) == index.end()) {
|
|
index[f.timestamp_rtp] = static_cast<int>(superframes.size());
|
|
superframes.push_back(f);
|
|
continue;
|
|
}
|
|
|
|
Frame& sf = superframes[index[f.timestamp_rtp]];
|
|
|
|
sf.width = std::max(sf.width, f.width);
|
|
sf.height = std::max(sf.height, f.height);
|
|
sf.frame_size += f.frame_size;
|
|
sf.keyframe |= f.keyframe;
|
|
|
|
sf.encode_time = std::max(sf.encode_time, f.encode_time);
|
|
sf.decode_time = std::max(sf.decode_time, f.decode_time);
|
|
|
|
if (f.spatial_idx > sf.spatial_idx) {
|
|
if (f.qp) {
|
|
sf.qp = f.qp;
|
|
}
|
|
if (f.psnr) {
|
|
sf.psnr = f.psnr;
|
|
}
|
|
}
|
|
|
|
sf.spatial_idx = std::max(sf.spatial_idx, f.spatial_idx);
|
|
sf.temporal_idx = std::max(sf.temporal_idx, f.temporal_idx);
|
|
|
|
sf.encoded |= f.encoded;
|
|
sf.decoded |= f.decoded;
|
|
}
|
|
|
|
return superframes;
|
|
}
|
|
|
|
Timestamp RtpToTime(uint32_t timestamp_rtp) {
|
|
return Timestamp::Micros((timestamp_rtp / k90kHz).us());
|
|
}
|
|
|
|
SamplesStatsCounter::StatsSample StatsSample(double value, Timestamp time) {
|
|
return SamplesStatsCounter::StatsSample{value, time};
|
|
}
|
|
|
|
TimeDelta CalcTotalDuration(const std::vector<Frame>& frames) {
|
|
RTC_CHECK(!frames.empty());
|
|
TimeDelta duration = TimeDelta::Zero();
|
|
if (frames.size() > 1) {
|
|
duration +=
|
|
(frames.rbegin()->timestamp_rtp - frames.begin()->timestamp_rtp) /
|
|
k90kHz;
|
|
}
|
|
|
|
// Add last frame duration. If target frame rate is provided, calculate frame
|
|
// duration from it. Otherwise, assume duration of last frame is the same as
|
|
// duration of preceding frame.
|
|
if (frames.rbegin()->target_framerate) {
|
|
duration += 1 / *frames.rbegin()->target_framerate;
|
|
} else {
|
|
RTC_CHECK_GT(frames.size(), 1u);
|
|
duration += (frames.rbegin()->timestamp_rtp -
|
|
std::next(frames.rbegin())->timestamp_rtp) /
|
|
k90kHz;
|
|
}
|
|
|
|
return duration;
|
|
}
|
|
} // namespace
|
|
|
|
std::vector<Frame> VideoCodecStatsImpl::Slice(
|
|
absl::optional<Filter> filter) const {
|
|
std::vector<Frame> frames;
|
|
for (const auto& [frame_id, f] : frames_) {
|
|
if (filter.has_value()) {
|
|
if (filter->first_frame.has_value() &&
|
|
f.frame_num < *filter->first_frame) {
|
|
continue;
|
|
}
|
|
if (filter->last_frame.has_value() && f.frame_num > *filter->last_frame) {
|
|
continue;
|
|
}
|
|
if (filter->spatial_idx.has_value() &&
|
|
f.spatial_idx != *filter->spatial_idx) {
|
|
continue;
|
|
}
|
|
if (filter->temporal_idx.has_value() &&
|
|
f.temporal_idx > *filter->temporal_idx) {
|
|
continue;
|
|
}
|
|
}
|
|
frames.push_back(f);
|
|
}
|
|
return frames;
|
|
}
|
|
|
|
Stream VideoCodecStatsImpl::Aggregate(const std::vector<Frame>& frames) const {
|
|
std::vector<Frame> superframes = Merge(frames);
|
|
RTC_CHECK(!superframes.empty());
|
|
|
|
LeakyBucket leacky_bucket;
|
|
Stream stream;
|
|
for (size_t i = 0; i < superframes.size(); ++i) {
|
|
Frame& f = superframes[i];
|
|
Timestamp time = RtpToTime(f.timestamp_rtp);
|
|
|
|
if (!f.frame_size.IsZero()) {
|
|
stream.width.AddSample(StatsSample(f.width, time));
|
|
stream.height.AddSample(StatsSample(f.height, time));
|
|
stream.frame_size_bytes.AddSample(
|
|
StatsSample(f.frame_size.bytes(), time));
|
|
stream.keyframe.AddSample(StatsSample(f.keyframe, time));
|
|
if (f.qp) {
|
|
stream.qp.AddSample(StatsSample(*f.qp, time));
|
|
}
|
|
}
|
|
|
|
if (f.encoded) {
|
|
stream.encode_time_ms.AddSample(StatsSample(f.encode_time.ms(), time));
|
|
}
|
|
|
|
if (f.decoded) {
|
|
stream.decode_time_ms.AddSample(StatsSample(f.decode_time.ms(), time));
|
|
}
|
|
|
|
if (f.psnr) {
|
|
stream.psnr.y.AddSample(StatsSample(f.psnr->y, time));
|
|
stream.psnr.u.AddSample(StatsSample(f.psnr->u, time));
|
|
stream.psnr.v.AddSample(StatsSample(f.psnr->v, time));
|
|
}
|
|
|
|
if (f.target_framerate) {
|
|
stream.target_framerate_fps.AddSample(
|
|
StatsSample(f.target_framerate->millihertz() / 1000.0, time));
|
|
}
|
|
|
|
if (f.target_bitrate) {
|
|
stream.target_bitrate_kbps.AddSample(
|
|
StatsSample(f.target_bitrate->bps() / 1000.0, time));
|
|
|
|
int buffer_level_bits = leacky_bucket.Update(f);
|
|
stream.transmission_time_ms.AddSample(
|
|
StatsSample(buffer_level_bits * rtc::kNumMillisecsPerSec /
|
|
f.target_bitrate->bps(),
|
|
RtpToTime(f.timestamp_rtp)));
|
|
}
|
|
}
|
|
|
|
TimeDelta duration = CalcTotalDuration(superframes);
|
|
DataRate encoded_bitrate =
|
|
DataSize::Bytes(stream.frame_size_bytes.GetSum()) / duration;
|
|
|
|
int num_encoded_frames = stream.frame_size_bytes.NumSamples();
|
|
Frequency encoded_framerate = num_encoded_frames / duration;
|
|
|
|
absl::optional<double> bitrate_mismatch_pct;
|
|
if (auto target_bitrate = superframes.begin()->target_bitrate;
|
|
target_bitrate) {
|
|
bitrate_mismatch_pct = 100.0 *
|
|
(encoded_bitrate.bps() - target_bitrate->bps()) /
|
|
target_bitrate->bps();
|
|
}
|
|
|
|
absl::optional<double> framerate_mismatch_pct;
|
|
if (auto target_framerate = superframes.begin()->target_framerate;
|
|
target_framerate) {
|
|
framerate_mismatch_pct =
|
|
100.0 *
|
|
(encoded_framerate.millihertz() - target_framerate->millihertz()) /
|
|
target_framerate->millihertz();
|
|
}
|
|
|
|
for (auto& f : superframes) {
|
|
Timestamp time = RtpToTime(f.timestamp_rtp);
|
|
stream.encoded_bitrate_kbps.AddSample(
|
|
StatsSample(encoded_bitrate.bps() / 1000.0, time));
|
|
|
|
stream.encoded_framerate_fps.AddSample(
|
|
StatsSample(encoded_framerate.millihertz() / 1000.0, time));
|
|
|
|
if (bitrate_mismatch_pct) {
|
|
stream.bitrate_mismatch_pct.AddSample(
|
|
StatsSample(*bitrate_mismatch_pct, time));
|
|
}
|
|
|
|
if (framerate_mismatch_pct) {
|
|
stream.framerate_mismatch_pct.AddSample(
|
|
StatsSample(*framerate_mismatch_pct, time));
|
|
}
|
|
}
|
|
|
|
return stream;
|
|
}
|
|
|
|
void VideoCodecStatsImpl::AddFrame(const Frame& frame) {
|
|
FrameId frame_id{.timestamp_rtp = frame.timestamp_rtp,
|
|
.spatial_idx = frame.spatial_idx};
|
|
RTC_CHECK(frames_.find(frame_id) == frames_.end())
|
|
<< "Frame with timestamp_rtp=" << frame.timestamp_rtp
|
|
<< " and spatial_idx=" << frame.spatial_idx << " already exists";
|
|
|
|
frames_[frame_id] = frame;
|
|
}
|
|
|
|
Frame* VideoCodecStatsImpl::GetFrame(uint32_t timestamp_rtp, int spatial_idx) {
|
|
FrameId frame_id{.timestamp_rtp = timestamp_rtp, .spatial_idx = spatial_idx};
|
|
if (frames_.find(frame_id) == frames_.end()) {
|
|
return nullptr;
|
|
}
|
|
return &frames_.find(frame_id)->second;
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|