mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00

This is a reland of 1880c7162b
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org, stefan@webrtc.org
Bug: webrtc:8524
Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae
Reviewed-on: https://webrtc-review.googlesource.com/40202
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21668}
130 lines
4.5 KiB
C++
130 lines
4.5 KiB
C++
/*
|
|
* Copyright (c) 2017 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 MODULES_VIDEO_CODING_CODECS_TEST_TEST_CONFIG_H_
|
|
#define MODULES_VIDEO_CODING_CODECS_TEST_TEST_CONFIG_H_
|
|
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "common_types.h" // NOLINT(build/include)
|
|
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
|
|
#include "modules/video_coding/codecs/test/packet_manipulator.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
// Defines which frame types shall be excluded from packet loss and when.
|
|
enum ExcludeFrameTypes {
|
|
// Will exclude the first keyframe in the video sequence from packet loss.
|
|
// Following keyframes will be targeted for packet loss.
|
|
kExcludeOnlyFirstKeyFrame,
|
|
// Exclude all keyframes from packet loss, no matter where in the video
|
|
// sequence they occur.
|
|
kExcludeAllKeyFrames
|
|
};
|
|
|
|
// Test configuration for a test run.
|
|
struct TestConfig {
|
|
class EncodedFrameChecker {
|
|
public:
|
|
virtual ~EncodedFrameChecker() = default;
|
|
|
|
virtual void CheckEncodedFrame(webrtc::VideoCodecType codec,
|
|
const EncodedImage& encoded_frame) const = 0;
|
|
};
|
|
|
|
void SetCodecSettings(VideoCodecType codec_type,
|
|
size_t num_temporal_layers,
|
|
bool error_concealment_on,
|
|
bool denoising_on,
|
|
bool frame_dropper_on,
|
|
bool spatial_resize_on,
|
|
bool resilience_on,
|
|
size_t width,
|
|
size_t height);
|
|
|
|
size_t NumberOfCores() const;
|
|
size_t NumberOfTemporalLayers() const;
|
|
size_t NumberOfSpatialLayers() const;
|
|
size_t TemporalLayerForFrame(size_t frame_idx) const;
|
|
std::vector<FrameType> FrameTypeForFrame(size_t frame_idx) const;
|
|
std::string ToString() const;
|
|
std::string CodecName() const;
|
|
std::string FilenameWithParams() const;
|
|
|
|
// Plain name of YUV file to process without file extension.
|
|
std::string filename;
|
|
|
|
// File to process. This must be a video file in the YUV format.
|
|
std::string input_filename;
|
|
|
|
// File to write to during processing for the test. Will be a video file in
|
|
// the YUV format.
|
|
std::string output_filename;
|
|
|
|
// Number of frames to process.
|
|
size_t num_frames = 0;
|
|
|
|
// Configurations related to networking.
|
|
NetworkingConfig networking_config;
|
|
|
|
// Decides how the packet loss simulations shall exclude certain frames from
|
|
// packet loss.
|
|
ExcludeFrameTypes exclude_frame_types = kExcludeOnlyFirstKeyFrame;
|
|
|
|
// Force the encoder and decoder to use a single core for processing.
|
|
// Using a single core is necessary to get a deterministic behavior for the
|
|
// encoded frames - using multiple cores will produce different encoded frames
|
|
// since multiple cores are competing to consume the byte budget for each
|
|
// frame in parallel.
|
|
// If set to false, the maximum number of available cores will be used.
|
|
bool use_single_core = false;
|
|
|
|
// Should cpu usage be measured?
|
|
// If set to true, the encoding will run in real-time.
|
|
bool measure_cpu = false;
|
|
|
|
// If > 0: forces the encoder to create a keyframe every Nth frame.
|
|
// Note that the encoder may create a keyframe in other locations in addition
|
|
// to this setting. Forcing key frames may also affect encoder planning
|
|
// optimizations in a negative way, since it will suddenly be forced to
|
|
// produce an expensive key frame.
|
|
size_t keyframe_interval = 0;
|
|
|
|
// Codec settings to use.
|
|
webrtc::VideoCodec codec_settings;
|
|
|
|
// H.264 specific settings.
|
|
struct H264CodecSettings {
|
|
H264::Profile profile = H264::kProfileConstrainedBaseline;
|
|
H264PacketizationMode packetization_mode =
|
|
webrtc::H264PacketizationMode::NonInterleaved;
|
|
} h264_codec_settings;
|
|
|
|
// Should hardware accelerated codecs be used?
|
|
bool hw_encoder = false;
|
|
bool hw_decoder = false;
|
|
|
|
// Should the hardware codecs be wrapped in software fallbacks?
|
|
bool sw_fallback_encoder = false;
|
|
bool sw_fallback_decoder = false;
|
|
|
|
// Custom checker that will be called for each frame.
|
|
const EncodedFrameChecker* encoded_frame_checker = nullptr;
|
|
|
|
// Print out frame level stats.
|
|
bool print_frame_level_stats = false;
|
|
};
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_VIDEO_CODING_CODECS_TEST_TEST_CONFIG_H_
|