Source code

Revision control

Copy as Markdown

Other Tools

/*
* Copyright (c) 2022 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_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINK_H_
#define TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINK_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/test/metrics/metrics_logger.h"
#include "api/test/pclf/media_configuration.h"
#include "api/test/video/video_frame_writer.h"
#include "api/test/video_quality_analyzer_interface.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
#include "test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h"
namespace webrtc {
namespace webrtc_pc_e2e {
// A sink to inject video quality analyzer as a sink into WebRTC.
class AnalyzingVideoSink : public rtc::VideoSinkInterface<VideoFrame> {
public:
struct Stats {
// Time required to scale video frame to the requested rendered resolution.
// Collected only for frames with ID set and iff `report_infra_stats` is
// true.
SamplesStatsCounter scaling_tims_ms;
// Time required to process single video frame. Collected only for frames
// with ID set and iff `report_infra_stats` is true.
SamplesStatsCounter analyzing_sink_processing_time_ms;
};
AnalyzingVideoSink(absl::string_view peer_name,
Clock* clock,
VideoQualityAnalyzerInterface& analyzer,
AnalyzingVideoSinksHelper& sinks_helper,
const VideoSubscription& subscription,
bool report_infra_stats);
// Updates subscription used by this peer to render received video.
void UpdateSubscription(const VideoSubscription& subscription);
void OnFrame(const VideoFrame& frame) override;
void LogMetrics(webrtc::test::MetricsLogger& metrics_logger,
absl::string_view test_case_name) const;
Stats stats() const;
private:
struct SinksDescriptor {
SinksDescriptor(absl::string_view sender_peer_name,
const VideoResolution& resolution)
: sender_peer_name(sender_peer_name), resolution(resolution) {}
// Required to be able to resolve resolutions on new subscription and
// understand if we need to recreate `video_frame_writer` and `sinks`.
std::string sender_peer_name;
// Resolution which was used to create `video_frame_writer` and `sinks`.
VideoResolution resolution;
// Is set if dumping of output video was requested;
test::VideoFrameWriter* video_frame_writer = nullptr;
std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
};
// Scales video frame to `required_resolution` if necessary. Crashes if video
// frame and `required_resolution` have different aspect ratio.
VideoFrame ScaleVideoFrame(const VideoFrame& frame,
const VideoResolution& required_resolution)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Creates full copy of the frame to free any frame owned internal buffers
// and passes created copy to analyzer. Uses `I420Buffer` to represent
// frame content.
void AnalyzeFrame(const VideoFrame& frame);
// Populates sink for specified stream and caches them in `stream_sinks_`.
SinksDescriptor* PopulateSinks(absl::string_view stream_label)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
const std::string peer_name_;
const bool report_infra_stats_;
Clock* const clock_;
VideoQualityAnalyzerInterface* const analyzer_;
AnalyzingVideoSinksHelper* const sinks_helper_;
mutable Mutex mutex_;
VideoSubscription subscription_ RTC_GUARDED_BY(mutex_);
std::map<std::string, SinksDescriptor> stream_sinks_ RTC_GUARDED_BY(mutex_);
Stats stats_ RTC_GUARDED_BY(mutex_);
};
} // namespace webrtc_pc_e2e
} // namespace webrtc
#endif // TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINK_H_