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This feature is not needed in video codec testing framework. In WebRTC video codecs never deal with packet loss. Packet loss is handled by jitter buffer which prevents passing of incomplete frames to decoder. Bug: webrtc:8768 Change-Id: I211cf51d913bec6a1f935e30691661d428ebd3b6 Reviewed-on: https://webrtc-review.googlesource.com/40740 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21722}
310 lines
12 KiB
C++
310 lines
12 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/video_coding/codecs/test/videoprocessor.h"
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#include <algorithm>
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#include <limits>
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#include <utility>
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#include "api/video/i420_buffer.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "common_video/h264/h264_common.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
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#include "modules/video_coding/include/video_codec_initializer.h"
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#include "modules/video_coding/utility/default_video_bitrate_allocator.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/timeutils.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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namespace {
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std::unique_ptr<VideoBitrateAllocator> CreateBitrateAllocator(
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TestConfig* config) {
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std::unique_ptr<TemporalLayersFactory> tl_factory;
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if (config->codec_settings.codecType == VideoCodecType::kVideoCodecVP8) {
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tl_factory.reset(new TemporalLayersFactory());
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config->codec_settings.VP8()->tl_factory = tl_factory.get();
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}
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return std::unique_ptr<VideoBitrateAllocator>(
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VideoCodecInitializer::CreateBitrateAllocator(config->codec_settings,
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std::move(tl_factory)));
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}
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size_t GetMaxNaluSizeBytes(const EncodedImage& encoded_frame,
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const TestConfig& config) {
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if (config.codec_settings.codecType != kVideoCodecH264)
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return 0;
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std::vector<webrtc::H264::NaluIndex> nalu_indices =
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webrtc::H264::FindNaluIndices(encoded_frame._buffer,
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encoded_frame._length);
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RTC_CHECK(!nalu_indices.empty());
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size_t max_size = 0;
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for (const webrtc::H264::NaluIndex& index : nalu_indices)
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max_size = std::max(max_size, index.payload_size);
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return max_size;
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}
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int GetElapsedTimeMicroseconds(int64_t start_ns, int64_t stop_ns) {
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int64_t diff_us = (stop_ns - start_ns) / rtc::kNumNanosecsPerMicrosec;
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RTC_DCHECK_GE(diff_us, std::numeric_limits<int>::min());
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RTC_DCHECK_LE(diff_us, std::numeric_limits<int>::max());
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return static_cast<int>(diff_us);
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}
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void ExtractBufferWithSize(const VideoFrame& image,
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int width,
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int height,
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rtc::Buffer* buffer) {
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if (image.width() != width || image.height() != height) {
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EXPECT_DOUBLE_EQ(static_cast<double>(width) / height,
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static_cast<double>(image.width()) / image.height());
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// Same aspect ratio, no cropping needed.
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rtc::scoped_refptr<I420Buffer> scaled(I420Buffer::Create(width, height));
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scaled->ScaleFrom(*image.video_frame_buffer()->ToI420());
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size_t length =
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CalcBufferSize(VideoType::kI420, scaled->width(), scaled->height());
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buffer->SetSize(length);
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RTC_CHECK_NE(ExtractBuffer(scaled, length, buffer->data()), -1);
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return;
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}
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// No resize.
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size_t length =
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CalcBufferSize(VideoType::kI420, image.width(), image.height());
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buffer->SetSize(length);
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RTC_CHECK_NE(ExtractBuffer(image, length, buffer->data()), -1);
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}
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} // namespace
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VideoProcessor::VideoProcessor(webrtc::VideoEncoder* encoder,
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webrtc::VideoDecoder* decoder,
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FrameReader* analysis_frame_reader,
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const TestConfig& config,
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Stats* stats,
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IvfFileWriter* encoded_frame_writer,
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FrameWriter* decoded_frame_writer)
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: config_(config),
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encoder_(encoder),
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decoder_(decoder),
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bitrate_allocator_(CreateBitrateAllocator(&config_)),
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encode_callback_(this),
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decode_callback_(this),
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analysis_frame_reader_(analysis_frame_reader),
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encoded_frame_writer_(encoded_frame_writer),
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decoded_frame_writer_(decoded_frame_writer),
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last_inputed_frame_num_(0),
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last_encoded_frame_num_(0),
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last_decoded_frame_num_(0),
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num_encoded_frames_(0),
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num_decoded_frames_(0),
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last_decoded_frame_buffer_(analysis_frame_reader->FrameLength()),
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stats_(stats) {
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RTC_DCHECK(encoder);
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RTC_DCHECK(decoder);
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RTC_DCHECK(analysis_frame_reader);
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RTC_DCHECK(stats);
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// Setup required callbacks for the encoder and decoder.
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RTC_CHECK_EQ(encoder_->RegisterEncodeCompleteCallback(&encode_callback_),
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WEBRTC_VIDEO_CODEC_OK);
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RTC_CHECK_EQ(decoder_->RegisterDecodeCompleteCallback(&decode_callback_),
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WEBRTC_VIDEO_CODEC_OK);
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// Initialize the encoder and decoder.
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RTC_CHECK_EQ(encoder_->InitEncode(&config_.codec_settings,
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static_cast<int>(config_.NumberOfCores()),
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config_.max_payload_size_bytes),
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WEBRTC_VIDEO_CODEC_OK);
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RTC_CHECK_EQ(decoder_->InitDecode(&config_.codec_settings,
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static_cast<int>(config_.NumberOfCores())),
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WEBRTC_VIDEO_CODEC_OK);
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}
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VideoProcessor::~VideoProcessor() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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RTC_CHECK_EQ(encoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
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RTC_CHECK_EQ(decoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
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encoder_->RegisterEncodeCompleteCallback(nullptr);
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decoder_->RegisterDecodeCompleteCallback(nullptr);
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}
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void VideoProcessor::ProcessFrame() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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const size_t frame_number = last_inputed_frame_num_++;
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// Get frame from file.
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rtc::scoped_refptr<I420BufferInterface> buffer(
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analysis_frame_reader_->ReadFrame());
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RTC_CHECK(buffer) << "Tried to read too many frames from the file.";
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// Use the frame number as the basis for timestamp to identify frames. Let the
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// first timestamp be non-zero, to not make the IvfFileWriter believe that we
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// want to use capture timestamps in the IVF files.
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// TODO(asapersson): Time stamps jump back if framerate increases.
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const size_t rtp_timestamp = (frame_number + 1) * kVideoPayloadTypeFrequency /
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config_.codec_settings.maxFramerate;
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const int64_t render_time_ms = (frame_number + 1) * rtc::kNumMillisecsPerSec /
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config_.codec_settings.maxFramerate;
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input_frames_[frame_number] =
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rtc::MakeUnique<VideoFrame>(buffer, static_cast<uint32_t>(rtp_timestamp),
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render_time_ms, webrtc::kVideoRotation_0);
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std::vector<FrameType> frame_types = config_.FrameTypeForFrame(frame_number);
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// Create frame statistics object used for aggregation at end of test run.
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FrameStatistic* frame_stat = stats_->AddFrame(rtp_timestamp);
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// For the highest measurement accuracy of the encode time, the start/stop
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// time recordings should wrap the Encode call as tightly as possible.
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frame_stat->encode_start_ns = rtc::TimeNanos();
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frame_stat->encode_return_code =
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encoder_->Encode(*input_frames_[frame_number], nullptr, &frame_types);
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}
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void VideoProcessor::SetRates(size_t bitrate_kbps, size_t framerate_fps) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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config_.codec_settings.maxFramerate = static_cast<uint32_t>(framerate_fps);
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bitrate_allocation_ = bitrate_allocator_->GetAllocation(
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static_cast<uint32_t>(bitrate_kbps * 1000),
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static_cast<uint32_t>(framerate_fps));
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const int set_rates_result = encoder_->SetRateAllocation(
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bitrate_allocation_, static_cast<uint32_t>(framerate_fps));
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RTC_DCHECK_GE(set_rates_result, 0)
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<< "Failed to update encoder with new rate " << bitrate_kbps << ".";
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}
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void VideoProcessor::FrameEncoded(webrtc::VideoCodecType codec,
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const EncodedImage& encoded_image) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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// For the highest measurement accuracy of the encode time, the start/stop
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// time recordings should wrap the Encode call as tightly as possible.
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int64_t encode_stop_ns = rtc::TimeNanos();
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if (config_.encoded_frame_checker) {
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config_.encoded_frame_checker->CheckEncodedFrame(codec, encoded_image);
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}
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FrameStatistic* frame_stat =
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stats_->GetFrameWithTimestamp(encoded_image._timeStamp);
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// Ensure strict monotonicity.
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const size_t frame_number = frame_stat->frame_number;
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if (num_encoded_frames_ > 0) {
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RTC_CHECK_GT(frame_number, last_encoded_frame_num_);
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}
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last_encoded_frame_num_ = frame_number;
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// Update frame statistics.
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frame_stat->encode_time_us =
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GetElapsedTimeMicroseconds(frame_stat->encode_start_ns, encode_stop_ns);
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frame_stat->encoding_successful = true;
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frame_stat->encoded_frame_size_bytes = encoded_image._length;
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frame_stat->frame_type = encoded_image._frameType;
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frame_stat->temporal_layer_idx = config_.TemporalLayerForFrame(frame_number);
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frame_stat->qp = encoded_image.qp_;
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frame_stat->target_bitrate_kbps =
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bitrate_allocation_.GetSpatialLayerSum(0) / 1000;
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frame_stat->max_nalu_size_bytes = GetMaxNaluSizeBytes(encoded_image, config_);
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// For the highest measurement accuracy of the decode time, the start/stop
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// time recordings should wrap the Decode call as tightly as possible.
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frame_stat->decode_start_ns = rtc::TimeNanos();
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frame_stat->decode_return_code =
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decoder_->Decode(encoded_image, false, nullptr);
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if (encoded_frame_writer_) {
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RTC_CHECK(encoded_frame_writer_->WriteFrame(encoded_image, codec));
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}
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++num_encoded_frames_;
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}
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void VideoProcessor::FrameDecoded(const VideoFrame& decoded_frame) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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// For the highest measurement accuracy of the decode time, the start/stop
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// time recordings should wrap the Decode call as tightly as possible.
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int64_t decode_stop_ns = rtc::TimeNanos();
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// Update frame statistics.
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FrameStatistic* frame_stat =
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stats_->GetFrameWithTimestamp(decoded_frame.timestamp());
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frame_stat->decoded_width = decoded_frame.width();
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frame_stat->decoded_height = decoded_frame.height();
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frame_stat->decode_time_us =
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GetElapsedTimeMicroseconds(frame_stat->decode_start_ns, decode_stop_ns);
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frame_stat->decoding_successful = true;
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// Ensure strict monotonicity.
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const size_t frame_number = frame_stat->frame_number;
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if (num_decoded_frames_ > 0) {
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RTC_CHECK_GT(frame_number, last_decoded_frame_num_);
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}
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// Check if the codecs have resized the frame since previously decoded frame.
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if (frame_number > 0) {
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if (decoded_frame_writer_ && num_decoded_frames_ > 0) {
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// For dropped/lost frames, write out the last decoded frame to make it
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// look like a freeze at playback.
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const size_t num_dropped_frames =
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frame_number - last_decoded_frame_num_ - 1;
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for (size_t i = 0; i < num_dropped_frames; i++) {
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WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
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}
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}
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}
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last_decoded_frame_num_ = frame_number;
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// Skip quality metrics calculation to not affect CPU usage.
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if (!config_.measure_cpu) {
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frame_stat->psnr =
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I420PSNR(input_frames_[frame_number].get(), &decoded_frame);
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frame_stat->ssim =
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I420SSIM(input_frames_[frame_number].get(), &decoded_frame);
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}
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// Delay erasing of input frames by one frame. The current frame might
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// still be needed for other simulcast stream or spatial layer.
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if (frame_number > 0) {
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auto input_frame_erase_to = input_frames_.lower_bound(frame_number - 1);
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input_frames_.erase(input_frames_.begin(), input_frame_erase_to);
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}
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if (decoded_frame_writer_) {
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ExtractBufferWithSize(decoded_frame, config_.codec_settings.width,
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config_.codec_settings.height,
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&last_decoded_frame_buffer_);
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WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
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}
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++num_decoded_frames_;
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}
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void VideoProcessor::WriteDecodedFrameToFile(rtc::Buffer* buffer) {
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RTC_DCHECK_EQ(buffer->size(), decoded_frame_writer_->FrameLength());
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RTC_CHECK(decoded_frame_writer_->WriteFrame(buffer->data()));
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}
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} // namespace test
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} // namespace webrtc
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