Files
platform-external-webrtc/modules/video_coding/codecs/test/videoprocessor.cc
Sergey Silkin 64eaa99cfc On-fly calculation of quality metrics.
Calculation of quality metrics required writing of decoded video
to file. There were two drawbacks with that approach. First, frame
drops significantly affected metrics because comparison was done
against the last decoded frame. Second, simulcast/SVC required
writing of multiple files. This might be too much data to dump.

On-fly metrics calculation is done in frame decoded callback.
Calculation time is excluded from encoding/decoding time. If CPU
usage measurement is enabled metrics calculation is disabled since
it affects CPU usage. The results are reported in Stats::PrintSummary.

Bug: webrtc:8524
Change-Id: Id54fb21f2f95deeb93757afaf46bde7d7ae18dac
Reviewed-on: https://webrtc-review.googlesource.com/22560
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20798}
2017-11-20 16:13:59 +00:00

377 lines
14 KiB
C++

/*
* 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.
*/
#include "modules/video_coding/codecs/test/videoprocessor.h"
#include <algorithm>
#include <limits>
#include <utility>
#include "api/video/i420_buffer.h"
#include "common_types.h" // NOLINT(build/include)
#include "common_video/h264/h264_common.h"
#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
#include "modules/video_coding/include/video_codec_initializer.h"
#include "modules/video_coding/utility/default_video_bitrate_allocator.h"
#include "rtc_base/checks.h"
#include "rtc_base/timeutils.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
namespace {
const int kRtpClockRateHz = 90000;
const int64_t kNoRenderTime = 0;
std::unique_ptr<VideoBitrateAllocator> CreateBitrateAllocator(
TestConfig* config) {
std::unique_ptr<TemporalLayersFactory> tl_factory;
if (config->codec_settings.codecType == VideoCodecType::kVideoCodecVP8) {
tl_factory.reset(new TemporalLayersFactory());
config->codec_settings.VP8()->tl_factory = tl_factory.get();
}
return std::unique_ptr<VideoBitrateAllocator>(
VideoCodecInitializer::CreateBitrateAllocator(config->codec_settings,
std::move(tl_factory)));
}
rtc::Optional<size_t> GetMaxNaluLength(const EncodedImage& encoded_frame,
const TestConfig& config) {
if (config.codec_settings.codecType != kVideoCodecH264)
return rtc::Optional<size_t>();
std::vector<webrtc::H264::NaluIndex> nalu_indices =
webrtc::H264::FindNaluIndices(encoded_frame._buffer,
encoded_frame._length);
RTC_CHECK(!nalu_indices.empty());
size_t max_length = 0;
for (const webrtc::H264::NaluIndex& index : nalu_indices)
max_length = std::max(max_length, index.payload_size);
return rtc::Optional<size_t>(max_length);
}
int GetElapsedTimeMicroseconds(int64_t start_ns, int64_t stop_ns) {
int64_t diff_us = (stop_ns - start_ns) / rtc::kNumNanosecsPerMicrosec;
RTC_DCHECK_GE(diff_us, std::numeric_limits<int>::min());
RTC_DCHECK_LE(diff_us, std::numeric_limits<int>::max());
return static_cast<int>(diff_us);
}
void ExtractBufferWithSize(const VideoFrame& image,
int width,
int height,
rtc::Buffer* buffer) {
if (image.width() != width || image.height() != height) {
EXPECT_DOUBLE_EQ(static_cast<double>(width) / height,
static_cast<double>(image.width()) / image.height());
// Same aspect ratio, no cropping needed.
rtc::scoped_refptr<I420Buffer> scaled(I420Buffer::Create(width, height));
scaled->ScaleFrom(*image.video_frame_buffer()->ToI420());
size_t length =
CalcBufferSize(VideoType::kI420, scaled->width(), scaled->height());
buffer->SetSize(length);
RTC_CHECK_NE(ExtractBuffer(scaled, length, buffer->data()), -1);
return;
}
// No resize.
size_t length =
CalcBufferSize(VideoType::kI420, image.width(), image.height());
buffer->SetSize(length);
RTC_CHECK_NE(ExtractBuffer(image, length, buffer->data()), -1);
}
} // namespace
VideoProcessor::VideoProcessor(webrtc::VideoEncoder* encoder,
webrtc::VideoDecoder* decoder,
FrameReader* analysis_frame_reader,
PacketManipulator* packet_manipulator,
const TestConfig& config,
Stats* stats,
IvfFileWriter* encoded_frame_writer,
FrameWriter* decoded_frame_writer)
: config_(config),
encoder_(encoder),
decoder_(decoder),
bitrate_allocator_(CreateBitrateAllocator(&config_)),
encode_callback_(this),
decode_callback_(this),
packet_manipulator_(packet_manipulator),
analysis_frame_reader_(analysis_frame_reader),
encoded_frame_writer_(encoded_frame_writer),
decoded_frame_writer_(decoded_frame_writer),
last_inputed_frame_num_(-1),
last_encoded_frame_num_(-1),
last_decoded_frame_num_(-1),
first_key_frame_has_been_excluded_(false),
last_decoded_frame_buffer_(analysis_frame_reader->FrameLength()),
stats_(stats),
rate_update_index_(-1) {
RTC_DCHECK(encoder);
RTC_DCHECK(decoder);
RTC_DCHECK(packet_manipulator);
RTC_DCHECK(analysis_frame_reader);
RTC_DCHECK(stats);
// Setup required callbacks for the encoder and decoder.
RTC_CHECK_EQ(encoder_->RegisterEncodeCompleteCallback(&encode_callback_),
WEBRTC_VIDEO_CODEC_OK);
RTC_CHECK_EQ(decoder_->RegisterDecodeCompleteCallback(&decode_callback_),
WEBRTC_VIDEO_CODEC_OK);
// Initialize the encoder and decoder.
RTC_CHECK_EQ(
encoder_->InitEncode(&config_.codec_settings, config_.NumberOfCores(),
config_.networking_config.max_payload_size_in_bytes),
WEBRTC_VIDEO_CODEC_OK);
RTC_CHECK_EQ(
decoder_->InitDecode(&config_.codec_settings, config_.NumberOfCores()),
WEBRTC_VIDEO_CODEC_OK);
}
VideoProcessor::~VideoProcessor() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
RTC_CHECK_EQ(encoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
RTC_CHECK_EQ(decoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
encoder_->RegisterEncodeCompleteCallback(nullptr);
decoder_->RegisterDecodeCompleteCallback(nullptr);
}
void VideoProcessor::ProcessFrame() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
const int frame_number = ++last_inputed_frame_num_;
// Get frame from file.
rtc::scoped_refptr<I420BufferInterface> buffer(
analysis_frame_reader_->ReadFrame());
RTC_CHECK(buffer) << "Tried to read too many frames from the file.";
// 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.
const uint32_t rtp_timestamp = (frame_number + 1) * kRtpClockRateHz /
config_.codec_settings.maxFramerate;
rtp_timestamp_to_frame_num_[rtp_timestamp] = frame_number;
input_frames_[frame_number] = rtc::MakeUnique<VideoFrame>(
buffer, rtp_timestamp, kNoRenderTime, webrtc::kVideoRotation_0);
std::vector<FrameType> frame_types = config_.FrameTypeForFrame(frame_number);
// Create frame statistics object used for aggregation at end of test run.
FrameStatistic* frame_stat = stats_->AddFrame();
// For the highest measurement accuracy of the encode time, the start/stop
// time recordings should wrap the Encode call as tightly as possible.
frame_stat->encode_start_ns = rtc::TimeNanos();
frame_stat->encode_return_code =
encoder_->Encode(*input_frames_[frame_number], nullptr, &frame_types);
}
void VideoProcessor::SetRates(int bitrate_kbps, int framerate_fps) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
config_.codec_settings.maxFramerate = framerate_fps;
int set_rates_result = encoder_->SetRateAllocation(
bitrate_allocator_->GetAllocation(bitrate_kbps * 1000, framerate_fps),
framerate_fps);
RTC_DCHECK_GE(set_rates_result, 0)
<< "Failed to update encoder with new rate " << bitrate_kbps << ".";
++rate_update_index_;
num_dropped_frames_.push_back(0);
num_spatial_resizes_.push_back(0);
}
std::vector<int> VideoProcessor::NumberDroppedFramesPerRateUpdate() const {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
return num_dropped_frames_;
}
std::vector<int> VideoProcessor::NumberSpatialResizesPerRateUpdate() const {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
return num_spatial_resizes_;
}
void VideoProcessor::FrameEncoded(webrtc::VideoCodecType codec,
const EncodedImage& encoded_image) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
// For the highest measurement accuracy of the encode time, the start/stop
// time recordings should wrap the Encode call as tightly as possible.
int64_t encode_stop_ns = rtc::TimeNanos();
if (config_.encoded_frame_checker) {
config_.encoded_frame_checker->CheckEncodedFrame(codec, encoded_image);
}
const int frame_number =
rtp_timestamp_to_frame_num_[encoded_image._timeStamp];
// Ensure strict monotonicity.
RTC_CHECK_GT(frame_number, last_encoded_frame_num_);
// Check for dropped frames.
bool last_frame_missing = false;
if (frame_number > 0) {
int num_dropped_from_last_encode =
frame_number - last_encoded_frame_num_ - 1;
RTC_DCHECK_GE(num_dropped_from_last_encode, 0);
RTC_CHECK_GE(rate_update_index_, 0);
num_dropped_frames_[rate_update_index_] += num_dropped_from_last_encode;
const FrameStatistic* last_encoded_frame_stat =
stats_->GetFrame(last_encoded_frame_num_);
last_frame_missing = (last_encoded_frame_stat->manipulated_length == 0);
}
last_encoded_frame_num_ = frame_number;
// Update frame statistics.
FrameStatistic* frame_stat = stats_->GetFrame(frame_number);
frame_stat->encode_time_us =
GetElapsedTimeMicroseconds(frame_stat->encode_start_ns, encode_stop_ns);
frame_stat->encoding_successful = true;
frame_stat->encoded_frame_size_bytes = encoded_image._length;
frame_stat->frame_type = encoded_image._frameType;
frame_stat->qp = encoded_image.qp_;
frame_stat->bitrate_kbps = static_cast<int>(
encoded_image._length * config_.codec_settings.maxFramerate * 8 / 1000);
frame_stat->total_packets =
encoded_image._length / config_.networking_config.packet_size_in_bytes +
1;
frame_stat->max_nalu_length = GetMaxNaluLength(encoded_image, config_);
// Make a raw copy of |encoded_image| to feed to the decoder.
size_t copied_buffer_size = encoded_image._length +
EncodedImage::GetBufferPaddingBytes(codec);
std::unique_ptr<uint8_t[]> copied_buffer(new uint8_t[copied_buffer_size]);
memcpy(copied_buffer.get(), encoded_image._buffer, encoded_image._length);
EncodedImage copied_image = encoded_image;
copied_image._size = copied_buffer_size;
copied_image._buffer = copied_buffer.get();
// Simulate packet loss.
if (!ExcludeFrame(copied_image)) {
frame_stat->packets_dropped =
packet_manipulator_->ManipulatePackets(&copied_image);
}
frame_stat->manipulated_length = copied_image._length;
// For the highest measurement accuracy of the decode time, the start/stop
// time recordings should wrap the Decode call as tightly as possible.
frame_stat->decode_start_ns = rtc::TimeNanos();
frame_stat->decode_return_code =
decoder_->Decode(copied_image, last_frame_missing, nullptr);
if (encoded_frame_writer_) {
RTC_CHECK(encoded_frame_writer_->WriteFrame(encoded_image, codec));
}
}
void VideoProcessor::FrameDecoded(const VideoFrame& decoded_frame) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
// For the highest measurement accuracy of the decode time, the start/stop
// time recordings should wrap the Decode call as tightly as possible.
int64_t decode_stop_ns = rtc::TimeNanos();
// Update frame statistics.
const int frame_number =
rtp_timestamp_to_frame_num_[decoded_frame.timestamp()];
FrameStatistic* frame_stat = stats_->GetFrame(frame_number);
frame_stat->decoded_width = decoded_frame.width();
frame_stat->decoded_height = decoded_frame.height();
frame_stat->decode_time_us =
GetElapsedTimeMicroseconds(frame_stat->decode_start_ns, decode_stop_ns);
frame_stat->decoding_successful = true;
// Ensure strict monotonicity.
RTC_CHECK_GT(frame_number, last_decoded_frame_num_);
// Check if the codecs have resized the frame since previously decoded frame.
if (frame_number > 0) {
if (decoded_frame_writer_ && last_decoded_frame_num_ >= 0) {
// For dropped/lost frames, write out the last decoded frame to make it
// look like a freeze at playback.
const int num_dropped_frames = frame_number - last_decoded_frame_num_;
for (int i = 0; i < num_dropped_frames; i++) {
WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
}
}
// TODO(ssilkin): move to FrameEncoded when webm:1474 is implemented.
const FrameStatistic* last_decoded_frame_stat =
stats_->GetFrame(last_decoded_frame_num_);
if (decoded_frame.width() != last_decoded_frame_stat->decoded_width ||
decoded_frame.height() != last_decoded_frame_stat->decoded_height) {
RTC_CHECK_GE(rate_update_index_, 0);
++num_spatial_resizes_[rate_update_index_];
}
}
last_decoded_frame_num_ = frame_number;
// Skip quality metrics calculation to not affect CPU usage.
if (!config_.measure_cpu) {
frame_stat->psnr =
I420PSNR(input_frames_[frame_number].get(), &decoded_frame);
frame_stat->ssim =
I420SSIM(input_frames_[frame_number].get(), &decoded_frame);
}
// Delay erasing of input frames by one frame. The current frame might
// still be needed for other simulcast stream or spatial layer.
const int frame_number_to_erase = frame_number - 1;
if (frame_number_to_erase >= 0) {
auto input_frame_erase_to =
input_frames_.lower_bound(frame_number_to_erase);
input_frames_.erase(input_frames_.begin(), input_frame_erase_to);
}
if (decoded_frame_writer_) {
ExtractBufferWithSize(decoded_frame, config_.codec_settings.width,
config_.codec_settings.height,
&last_decoded_frame_buffer_);
WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
}
}
void VideoProcessor::WriteDecodedFrameToFile(rtc::Buffer* buffer) {
RTC_DCHECK_EQ(buffer->size(), decoded_frame_writer_->FrameLength());
RTC_CHECK(decoded_frame_writer_->WriteFrame(buffer->data()));
}
bool VideoProcessor::ExcludeFrame(const EncodedImage& encoded_image) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
if (encoded_image._frameType != kVideoFrameKey) {
return false;
}
bool exclude_frame = false;
switch (config_.exclude_frame_types) {
case kExcludeOnlyFirstKeyFrame:
if (!first_key_frame_has_been_excluded_) {
first_key_frame_has_been_excluded_ = true;
exclude_frame = true;
}
break;
case kExcludeAllKeyFrames:
exclude_frame = true;
break;
default:
RTC_NOTREACHED();
}
return exclude_frame;
}
} // namespace test
} // namespace webrtc