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Add method FrameType for frame to TestConfig. Bug: none Change-Id: Icfeb12fcb961559c9b36a3aedb081a840b9d8556 Reviewed-on: https://webrtc-review.googlesource.com/16120 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20458}
372 lines
14 KiB
C++
372 lines
14 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/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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const int kRtpClockRateHz = 90000;
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const int64_t kNoRenderTime = 0;
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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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rtc::Optional<size_t> GetMaxNaluLength(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 rtc::Optional<size_t>();
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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_length = 0;
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for (const webrtc::H264::NaluIndex& index : nalu_indices)
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max_length = std::max(max_length, index.payload_size);
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return rtc::Optional<size_t>(max_length);
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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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FrameWriter* analysis_frame_writer,
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PacketManipulator* packet_manipulator,
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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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packet_manipulator_(packet_manipulator),
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analysis_frame_reader_(analysis_frame_reader),
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analysis_frame_writer_(analysis_frame_writer),
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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_(-1),
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last_encoded_frame_num_(-1),
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last_decoded_frame_num_(-1),
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first_key_frame_has_been_excluded_(false),
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last_decoded_frame_buffer_(analysis_frame_reader->FrameLength()),
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stats_(stats),
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rate_update_index_(-1) {
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RTC_DCHECK(encoder);
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RTC_DCHECK(decoder);
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RTC_DCHECK(packet_manipulator);
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RTC_DCHECK(analysis_frame_reader);
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RTC_DCHECK(analysis_frame_writer);
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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(
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encoder_->InitEncode(&config_.codec_settings, config_.NumberOfCores(),
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config_.networking_config.max_payload_size_in_bytes),
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WEBRTC_VIDEO_CODEC_OK);
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RTC_CHECK_EQ(
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decoder_->InitDecode(&config_.codec_settings, 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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++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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const uint32_t rtp_timestamp = (last_inputed_frame_num_ + 1) *
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kRtpClockRateHz /
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config_.codec_settings.maxFramerate;
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rtp_timestamp_to_frame_num_[rtp_timestamp] = last_inputed_frame_num_;
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VideoFrame source_frame(buffer, rtp_timestamp, kNoRenderTime,
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webrtc::kVideoRotation_0);
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std::vector<FrameType> frame_types =
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config_.FrameTypeForFrame(last_inputed_frame_num_);
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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();
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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(source_frame, nullptr, &frame_types);
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}
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void VideoProcessor::SetRates(int bitrate_kbps, int framerate_fps) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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config_.codec_settings.maxFramerate = framerate_fps;
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int set_rates_result = encoder_->SetRateAllocation(
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bitrate_allocator_->GetAllocation(bitrate_kbps * 1000, framerate_fps),
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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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++rate_update_index_;
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num_dropped_frames_.push_back(0);
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num_spatial_resizes_.push_back(0);
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}
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std::vector<int> VideoProcessor::NumberDroppedFramesPerRateUpdate() const {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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return num_dropped_frames_;
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}
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std::vector<int> VideoProcessor::NumberSpatialResizesPerRateUpdate() const {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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return num_spatial_resizes_;
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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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// Check for dropped frames.
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const int frame_number =
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rtp_timestamp_to_frame_num_[encoded_image._timeStamp];
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bool last_frame_missing = false;
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if (frame_number > 0) {
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RTC_DCHECK_GE(last_encoded_frame_num_, 0);
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int num_dropped_from_last_encode =
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frame_number - last_encoded_frame_num_ - 1;
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RTC_DCHECK_GE(num_dropped_from_last_encode, 0);
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RTC_CHECK_GE(rate_update_index_, 0);
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num_dropped_frames_[rate_update_index_] += num_dropped_from_last_encode;
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if (num_dropped_from_last_encode > 0) {
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// For dropped frames, we write out the last decoded frame to avoid
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// getting out of sync for the computation of PSNR and SSIM.
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for (int i = 0; i < num_dropped_from_last_encode; i++) {
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WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
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}
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}
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const FrameStatistic* last_encoded_frame_stat =
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stats_->GetFrame(last_encoded_frame_num_);
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last_frame_missing = (last_encoded_frame_stat->manipulated_length == 0);
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}
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// Ensure strict monotonicity.
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RTC_CHECK_GT(frame_number, last_encoded_frame_num_);
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last_encoded_frame_num_ = frame_number;
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// Update frame statistics.
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FrameStatistic* frame_stat = stats_->GetFrame(frame_number);
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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->qp = encoded_image.qp_;
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frame_stat->bitrate_kbps = static_cast<int>(
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encoded_image._length * config_.codec_settings.maxFramerate * 8 / 1000);
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frame_stat->total_packets =
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encoded_image._length / config_.networking_config.packet_size_in_bytes +
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1;
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frame_stat->max_nalu_length = GetMaxNaluLength(encoded_image, config_);
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// Make a raw copy of |encoded_image| to feed to the decoder.
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size_t copied_buffer_size = encoded_image._length +
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EncodedImage::GetBufferPaddingBytes(codec);
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std::unique_ptr<uint8_t[]> copied_buffer(new uint8_t[copied_buffer_size]);
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memcpy(copied_buffer.get(), encoded_image._buffer, encoded_image._length);
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EncodedImage copied_image = encoded_image;
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copied_image._size = copied_buffer_size;
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copied_image._buffer = copied_buffer.get();
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// Simulate packet loss.
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if (!ExcludeFrame(copied_image)) {
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frame_stat->packets_dropped =
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packet_manipulator_->ManipulatePackets(&copied_image);
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}
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frame_stat->manipulated_length = copied_image._length;
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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(copied_image, last_frame_missing, nullptr);
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if (frame_stat->decode_return_code != WEBRTC_VIDEO_CODEC_OK) {
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// Write the last successful frame the output file to avoid getting it out
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// of sync with the source file for SSIM and PSNR comparisons.
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WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
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}
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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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}
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void VideoProcessor::FrameDecoded(const VideoFrame& image) {
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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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const int frame_number = rtp_timestamp_to_frame_num_[image.timestamp()];
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FrameStatistic* frame_stat = stats_->GetFrame(frame_number);
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frame_stat->decoded_width = image.width();
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frame_stat->decoded_height = image.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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// Check if the codecs have resized the frame since previously decoded frame.
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if (frame_number > 0) {
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RTC_CHECK_GE(last_decoded_frame_num_, 0);
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const FrameStatistic* last_decoded_frame_stat =
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stats_->GetFrame(last_decoded_frame_num_);
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if (image.width() != last_decoded_frame_stat->decoded_width ||
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image.height() != last_decoded_frame_stat->decoded_height) {
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RTC_CHECK_GE(rate_update_index_, 0);
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++num_spatial_resizes_[rate_update_index_];
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}
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}
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// Ensure strict monotonicity.
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RTC_CHECK_GT(frame_number, last_decoded_frame_num_);
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last_decoded_frame_num_ = frame_number;
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// If the frame size is different from the original size, scale back to the
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// original size. This is needed for the PSNR and SSIM calculations.
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rtc::Buffer buffer;
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ExtractBufferWithSize(image, config_.codec_settings.width,
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config_.codec_settings.height, &buffer);
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WriteDecodedFrameToFile(&buffer);
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last_decoded_frame_buffer_ = std::move(buffer);
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}
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void VideoProcessor::WriteDecodedFrameToFile(rtc::Buffer* buffer) {
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RTC_DCHECK_EQ(buffer->size(), analysis_frame_writer_->FrameLength());
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RTC_CHECK(analysis_frame_writer_->WriteFrame(buffer->data()));
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if (decoded_frame_writer_) {
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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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}
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bool VideoProcessor::ExcludeFrame(const EncodedImage& encoded_image) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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if (encoded_image._frameType != kVideoFrameKey) {
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return false;
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}
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bool exclude_frame = false;
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switch (config_.exclude_frame_types) {
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case kExcludeOnlyFirstKeyFrame:
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if (!first_key_frame_has_been_excluded_) {
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first_key_frame_has_been_excluded_ = true;
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exclude_frame = true;
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}
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break;
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case kExcludeAllKeyFrames:
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exclude_frame = true;
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break;
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default:
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RTC_NOTREACHED();
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}
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return exclude_frame;
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}
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} // namespace test
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} // namespace webrtc
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