blob: 7c2c14f52f20b47c4767aed8cb601cec957874e9 [file] [log] [blame]
* Copyright (c) 2011 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 <string>
#include "modules/video_coding/codecs/interface/video_codec_interface.h"
#include "modules/video_coding/codecs/test/packet_manipulator.h"
#include "modules/video_coding/codecs/test/stats.h"
#include "system_wrappers/interface/tick_util.h"
#include "testsupport/frame_reader.h"
#include "testsupport/frame_writer.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.
// Exclude all keyframes from packet loss, no matter where in the video
// sequence they occur.
// Returns a string representation of the enum value.
const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e);
// Test configuration for a test run
struct TestConfig {
: name(""), description(""), test_number(0),
input_filename(""), output_filename(""), output_dir("out"),
networking_config(), exclude_frame_types(kExcludeOnlyFirstKeyFrame),
frame_length_in_bytes(-1), use_single_core(false), keyframe_interval(0),
codec_settings(NULL), verbose(true) {
// Name of the test. This is purely metadata and does not affect
// the test in any way.
std::string name;
// More detailed description of the test. This is purely metadata and does
// not affect the test in any way.
std::string description;
// Number of this test. Useful if multiple runs of the same test with
// different configurations shall be managed.
int test_number;
// File to process for the test. 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). Default: "out".
std::string output_dir;
// Configurations related to networking.
NetworkingConfig networking_config;
// Decides how the packet loss simulations shall exclude certain frames
// from packet loss. Default: kExcludeOnlyFirstKeyFrame.
ExcludeFrameTypes exclude_frame_types;
// The length of a single frame of the input video file. This value is
// calculated out of the width and height according to the video format
// specification. Must be set before processing.
int frame_length_in_bytes;
// 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.
// Default: false.
bool use_single_core;
// If set to a value >0 this setting forces the encoder to create a keyframe
// every Nth frame. Note that the encoder may create a keyframe in other
// locations in addition to the interval that is set using this parameter.
// 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.
// Default: 0.
int keyframe_interval;
// The codec settings to use for the test (target bitrate, video size,
// framerate and so on). This struct must be created and filled in using
// the VideoCodingModule::Codec() method.
webrtc::VideoCodec* codec_settings;
// If printing of information to stdout shall be performed during processing.
bool verbose;
// Returns a string representation of the enum value.
const char* VideoCodecTypeToStr(webrtc::VideoCodecType e);
// 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 {
virtual ~VideoProcessor() {}
// Performs initial calculations about frame size, sets up callbacks etc.
// Returns false if an error has occurred, in addition to printing to stderr.
virtual bool Init() = 0;
// Processes a single frame. Returns true as long as there's more frames
// available in the source clip.
// Frame number must be an integer >=0.
virtual bool ProcessFrame(int frame_number) = 0;
class VideoProcessorImpl : public VideoProcessor {
VideoProcessorImpl(webrtc::VideoEncoder* encoder,
webrtc::VideoDecoder* decoder,
FrameReader* frame_reader,
FrameWriter* frame_writer,
PacketManipulator* packet_manipulator,
const TestConfig& config,
Stats* stats);
virtual ~VideoProcessorImpl();
virtual bool Init();
virtual bool ProcessFrame(int frame_number);
// Invoked by the callback when a frame has completed encoding.
void FrameEncoded(EncodedImage* encodedImage);
// Invoked by the callback when a frame has completed decoding.
void FrameDecoded(const RawImage& image);
// Used for getting a 32-bit integer representing time
// (checks the size is within signed 32-bit bounds before casting it)
int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
const webrtc::TickTime& stop);
webrtc::VideoEncoder* encoder_;
webrtc::VideoDecoder* decoder_;
FrameReader* frame_reader_;
FrameWriter* frame_writer_;
PacketManipulator* packet_manipulator_;
const TestConfig& config_;
Stats* stats_;
EncodedImageCallback* encode_callback_;
DecodedImageCallback* decode_callback_;
// Buffer used for reading the source video file:
WebRtc_UWord8* source_buffer_;
// Keep track of the last successful frame, since we need to write that
// when decoding fails:
WebRtc_UWord8* last_successful_frame_buffer_;
webrtc::RawImage source_frame_;
// To keep track of if we have excluded the first key frame from packet loss:
bool first_key_frame_has_been_excluded_;
// To tell the decoder previous frame have been dropped due to packet loss:
bool last_frame_missing_;
// If Init() has executed successfully.
bool initialized_;
// Statistics
double bit_rate_factor_; // multiply frame length with this to get bit rate
webrtc::TickTime encode_start_;
webrtc::TickTime decode_start_;
// Callback class required to implement according to the VideoEncoder API.
class VideoProcessorEncodeCompleteCallback
: public webrtc::EncodedImageCallback {
explicit VideoProcessorEncodeCompleteCallback(VideoProcessorImpl* vp)
: video_processor_(vp) {
WebRtc_Word32 Encoded(
webrtc::EncodedImage& encoded_image,
const webrtc::CodecSpecificInfo* codec_specific_info = NULL,
const webrtc::RTPFragmentationHeader* fragmentation = NULL);
VideoProcessorImpl* video_processor_;
// Callback class required to implement according to the VideoDecoder API.
class VideoProcessorDecodeCompleteCallback
: public webrtc::DecodedImageCallback {
explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
: video_processor_(vp) {
WebRtc_Word32 Decoded(webrtc::RawImage& image);
VideoProcessorImpl* video_processor_;
} // namespace test
} // namespace webrtc