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/*
* Copyright (c) 2012 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 WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/video/video_frame.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
#include "webrtc/modules/video_coding/codecs/test/stats.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
Reland of Add optional visualization file writers to VideoProcessor tests. (patchset #1 id:1 of https://codereview.webrtc.org/2708103002/ ) Reason for revert: Necessary calls were "protected" by RTC_DCHECKs, that were optimized away in some release builds. Replacing the RTC_DCHECKs with EXPECTs. Original issue's description: > Revert of Add optional visualization file writers to VideoProcessor tests. (patchset #4 id:220001 of https://codereview.webrtc.org/2700493006/ ) > > Reason for revert: > Breaks downstream project. > > Original issue's description: > > Add optional visualization file writers to VideoProcessor tests. > > > > The purpose of this visualization CL is to add the ability to record > > video at the source, after encode, and after decode, in the VideoProcessor > > tests. These output files can then be replayed and used as a subjective > > complement to the objective metric plots given by the existing Python > > plotting script. > > > > BUG=webrtc:6634 > > > > Review-Url: https://codereview.webrtc.org/2700493006 > > Cr-Commit-Position: refs/heads/master@{#16738} > > Committed: https://chromium.googlesource.com/external/webrtc/+/872104ac41d7764f8676c9ea55555210bea4605c > > TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6634 > > Review-Url: https://codereview.webrtc.org/2708103002 > Cr-Commit-Position: refs/heads/master@{#16745} > Committed: https://chromium.googlesource.com/external/webrtc/+/2a8135a1741761bd6de52163c0dc35f6eff7c8eb TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2706123003 Cr-Commit-Position: refs/heads/master@{#16769}
2017-02-22 01:26:59 -08:00
#include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
#include "webrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/sequenced_task_checker.h"
#include "webrtc/rtc_base/task_queue.h"
#include "webrtc/test/testsupport/frame_reader.h"
#include "webrtc/test/testsupport/frame_writer.h"
namespace webrtc {
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
class VideoBitrateAllocator;
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
};
// Returns a string representation of the enum value.
const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e);
// Test configuration for a test run.
struct TestConfig {
// Name of the test. This is purely metadata and does not affect the test.
std::string name;
// More detailed description of the test. This is purely metadata and does
// not affect the test.
std::string description;
// Number of this test. Useful if multiple runs of the same test with
// different configurations shall be managed.
int test_number = 0;
// 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;
// Path to the directory where encoded files will be put
// (absolute or relative to the executable).
std::string output_dir = "out";
// 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;
// The length of a single frame of the input video file. Calculated out of the
// width and height according to the video format specification (i.e. YUV).
size_t frame_length_in_bytes = 0;
// 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;
// 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.
int keyframe_interval = 0;
// The codec settings to use for the test (target bitrate, video size,
// framerate and so on). This struct should be filled in using the
// VideoCodingModule::Codec() method.
webrtc::VideoCodec codec_settings;
// If printing of information to stdout shall be performed during processing.
bool verbose = true;
// If HW or SW codec should be used.
bool hw_codec = false;
// In batch mode, the VideoProcessor is fed all the frames for processing
// before any metrics are calculated. This is useful for pipelining HW codecs,
// for which some calculated metrics otherwise would be incorrect. The
// downside with batch mode is that mid-test rate allocation is not supported.
bool batch_mode = false;
};
// Handles encoding/decoding of video using the VideoEncoder/VideoDecoder
// interfaces. This is done in a sequential manner in order to be able to
// measure times properly.
// The class processes a frame at the time for the configured input file.
// It maintains state of where in the source input file the processing is at.
//
// Regarding packet loss: Note that keyframes are excluded (first or all
// depending on the ExcludeFrameTypes setting). This is because if key frames
// would be altered, all the following delta frames would be pretty much
// worthless. VP8 has an error-resilience feature that makes it able to handle
// packet loss in key non-first keyframes, which is why only the first is
// excluded by default.
// Packet loss in such important frames is handled on a higher level in the
// Video Engine, where signaling would request a retransmit of the lost packets,
// since they're so important.
//
// Note this class is not thread safe in any way and is meant for simple testing
// purposes.
class VideoProcessor {
public:
VideoProcessor(webrtc::VideoEncoder* encoder,
webrtc::VideoDecoder* decoder,
FrameReader* analysis_frame_reader,
FrameWriter* analysis_frame_writer,
PacketManipulator* packet_manipulator,
const TestConfig& config,
Stats* stats,
IvfFileWriter* encoded_frame_writer,
FrameWriter* decoded_frame_writer);
~VideoProcessor();
// Sets up callbacks and initializes the encoder and decoder.
void Init();
// Tears down callbacks and releases the encoder and decoder.
void Release();
// Processes a single frame. The frames must be processed in order, and the
// VideoProcessor must be initialized first.
void ProcessFrame(int frame_number);
// Updates the encoder with target rates. Must be called at least once.
void SetRates(int bitrate_kbps, int framerate_fps);
// TODO(brandtr): Get rid of these functions by moving the corresponding QP
// fields to the Stats object.
int GetQpFromEncoder(int frame_number) const;
int GetQpFromBitstream(int frame_number) const;
// Return the number of dropped frames.
int NumberDroppedFrames();
// Return the number of spatial resizes.
int NumberSpatialResizes();
private:
// Container that holds per-frame information that needs to be stored between
// calls to Encode and Decode, as well as the corresponding callbacks. It is
// not directly used for statistics -- for that, test::FrameStatistic is used.
// TODO(brandtr): Get rid of this struct and use the Stats class instead.
struct FrameInfo {
int64_t encode_start_ns = 0;
int64_t decode_start_ns = 0;
int qp_encoder = 0;
int qp_bitstream = 0;
int decoded_width = 0;
int decoded_height = 0;
size_t manipulated_length = 0;
};
class VideoProcessorEncodeCompleteCallback
: public webrtc::EncodedImageCallback {
public:
explicit VideoProcessorEncodeCompleteCallback(
VideoProcessor* video_processor)
: video_processor_(video_processor),
task_queue_(rtc::TaskQueue::Current()) {}
Result OnEncodedImage(
const webrtc::EncodedImage& encoded_image,
const webrtc::CodecSpecificInfo* codec_specific_info,
const webrtc::RTPFragmentationHeader* fragmentation) override {
RTC_CHECK(codec_specific_info);
if (task_queue_ && !task_queue_->IsCurrent()) {
task_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
new EncodeCallbackTask(video_processor_, encoded_image,
codec_specific_info, fragmentation)));
return Result(Result::OK, 0);
}
video_processor_->FrameEncoded(codec_specific_info->codecType,
encoded_image, fragmentation);
return Result(Result::OK, 0);
}
private:
class EncodeCallbackTask : public rtc::QueuedTask {
public:
EncodeCallbackTask(VideoProcessor* video_processor,
const webrtc::EncodedImage& encoded_image,
const webrtc::CodecSpecificInfo* codec_specific_info,
const webrtc::RTPFragmentationHeader* fragmentation)
: video_processor_(video_processor),
buffer_(encoded_image._buffer, encoded_image._length),
encoded_image_(encoded_image),
codec_specific_info_(*codec_specific_info) {
encoded_image_._buffer = buffer_.data();
RTC_CHECK(fragmentation);
fragmentation_.CopyFrom(*fragmentation);
}
bool Run() override {
video_processor_->FrameEncoded(codec_specific_info_.codecType,
encoded_image_, &fragmentation_);
return true;
}
private:
VideoProcessor* const video_processor_;
rtc::Buffer buffer_;
webrtc::EncodedImage encoded_image_;
const webrtc::CodecSpecificInfo codec_specific_info_;
webrtc::RTPFragmentationHeader fragmentation_;
};
VideoProcessor* const video_processor_;
rtc::TaskQueue* const task_queue_;
};
class VideoProcessorDecodeCompleteCallback
: public webrtc::DecodedImageCallback {
public:
explicit VideoProcessorDecodeCompleteCallback(
VideoProcessor* video_processor)
: video_processor_(video_processor),
task_queue_(rtc::TaskQueue::Current()) {}
int32_t Decoded(webrtc::VideoFrame& image) override {
if (task_queue_ && !task_queue_->IsCurrent()) {
task_queue_->PostTask(
[this, image]() { video_processor_->FrameDecoded(image); });
return 0;
}
video_processor_->FrameDecoded(image);
return 0;
}
int32_t Decoded(webrtc::VideoFrame& image,
int64_t decode_time_ms) override {
return Decoded(image);
}
void Decoded(webrtc::VideoFrame& image,
rtc::Optional<int32_t> decode_time_ms,
rtc::Optional<uint8_t> qp) override {
Decoded(image);
}
private:
VideoProcessor* const video_processor_;
rtc::TaskQueue* const task_queue_;
};
// Invoked by the callback adapter when a frame has completed encoding.
void FrameEncoded(webrtc::VideoCodecType codec,
const webrtc::EncodedImage& encodedImage,
const webrtc::RTPFragmentationHeader* fragmentation);
// Invoked by the callback adapter when a frame has completed decoding.
void FrameDecoded(const webrtc::VideoFrame& image);
// Use the frame number as the basis for timestamp to identify frames. Let the
// first timestamp be non-zero, to not make the IvfFileWriter believe that we
// want to use capture timestamps in the IVF files.
uint32_t FrameNumberToTimestamp(int frame_number) const;
int TimestampToFrameNumber(uint32_t timestamp) const;
bool initialized_ GUARDED_BY(sequence_checker_);
TestConfig config_ GUARDED_BY(sequence_checker_);
webrtc::VideoEncoder* const encoder_;
webrtc::VideoDecoder* const decoder_;
const std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_;
// Adapters for the codec callbacks.
VideoProcessorEncodeCompleteCallback encode_callback_;
VideoProcessorDecodeCompleteCallback decode_callback_;
// Fake network.
PacketManipulator* const packet_manipulator_;
Reland of Add optional visualization file writers to VideoProcessor tests. (patchset #1 id:1 of https://codereview.webrtc.org/2708103002/ ) Reason for revert: Necessary calls were "protected" by RTC_DCHECKs, that were optimized away in some release builds. Replacing the RTC_DCHECKs with EXPECTs. Original issue's description: > Revert of Add optional visualization file writers to VideoProcessor tests. (patchset #4 id:220001 of https://codereview.webrtc.org/2700493006/ ) > > Reason for revert: > Breaks downstream project. > > Original issue's description: > > Add optional visualization file writers to VideoProcessor tests. > > > > The purpose of this visualization CL is to add the ability to record > > video at the source, after encode, and after decode, in the VideoProcessor > > tests. These output files can then be replayed and used as a subjective > > complement to the objective metric plots given by the existing Python > > plotting script. > > > > BUG=webrtc:6634 > > > > Review-Url: https://codereview.webrtc.org/2700493006 > > Cr-Commit-Position: refs/heads/master@{#16738} > > Committed: https://chromium.googlesource.com/external/webrtc/+/872104ac41d7764f8676c9ea55555210bea4605c > > TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6634 > > Review-Url: https://codereview.webrtc.org/2708103002 > Cr-Commit-Position: refs/heads/master@{#16745} > Committed: https://chromium.googlesource.com/external/webrtc/+/2a8135a1741761bd6de52163c0dc35f6eff7c8eb TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2706123003 Cr-Commit-Position: refs/heads/master@{#16769}
2017-02-22 01:26:59 -08:00
// These (mandatory) file manipulators are used for, e.g., objective PSNR and
// SSIM calculations at the end of a test run.
FrameReader* const analysis_frame_reader_;
FrameWriter* const analysis_frame_writer_;
// These (optional) file writers are used to persistently store the encoded
// and decoded bitstreams. The purpose is to give the experimenter an option
// to subjectively evaluate the quality of the processing. Each frame writer
// is enabled by being non-null.
Reland of Add optional visualization file writers to VideoProcessor tests. (patchset #1 id:1 of https://codereview.webrtc.org/2708103002/ ) Reason for revert: Necessary calls were "protected" by RTC_DCHECKs, that were optimized away in some release builds. Replacing the RTC_DCHECKs with EXPECTs. Original issue's description: > Revert of Add optional visualization file writers to VideoProcessor tests. (patchset #4 id:220001 of https://codereview.webrtc.org/2700493006/ ) > > Reason for revert: > Breaks downstream project. > > Original issue's description: > > Add optional visualization file writers to VideoProcessor tests. > > > > The purpose of this visualization CL is to add the ability to record > > video at the source, after encode, and after decode, in the VideoProcessor > > tests. These output files can then be replayed and used as a subjective > > complement to the objective metric plots given by the existing Python > > plotting script. > > > > BUG=webrtc:6634 > > > > Review-Url: https://codereview.webrtc.org/2700493006 > > Cr-Commit-Position: refs/heads/master@{#16738} > > Committed: https://chromium.googlesource.com/external/webrtc/+/872104ac41d7764f8676c9ea55555210bea4605c > > TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6634 > > Review-Url: https://codereview.webrtc.org/2708103002 > Cr-Commit-Position: refs/heads/master@{#16745} > Committed: https://chromium.googlesource.com/external/webrtc/+/2a8135a1741761bd6de52163c0dc35f6eff7c8eb TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2706123003 Cr-Commit-Position: refs/heads/master@{#16769}
2017-02-22 01:26:59 -08:00
IvfFileWriter* const encoded_frame_writer_;
FrameWriter* const decoded_frame_writer_;
// Frame metadata for all frames that have been added through a call to
// ProcessFrames(). We need to store this metadata over the course of the
// test run, to support pipelining HW codecs.
std::vector<FrameInfo> frame_infos_ GUARDED_BY(sequence_checker_);
int last_encoded_frame_num_ GUARDED_BY(sequence_checker_);
int last_decoded_frame_num_ GUARDED_BY(sequence_checker_);
// Keep track of if we have excluded the first key frame from packet loss.
bool first_key_frame_has_been_excluded_ GUARDED_BY(sequence_checker_);
// Keep track of the last successfully decoded frame, since we write that
// frame to disk when decoding fails.
rtc::Buffer last_decoded_frame_buffer_ GUARDED_BY(sequence_checker_);
// Statistics.
Stats* stats_;
int num_dropped_frames_ GUARDED_BY(sequence_checker_);
int num_spatial_resizes_ GUARDED_BY(sequence_checker_);
rtc::SequencedTaskChecker sequence_checker_;
RTC_DISALLOW_COPY_AND_ASSIGN(VideoProcessor);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_