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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.
*/
#include "modules/video_coding/codecs/test/videoprocessor.h"
#include <string.h>
#include <algorithm>
#include <cstddef>
#include <limits>
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
#include <memory>
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 <utility>
#include "api/scoped_refptr.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video/video_frame_buffer.h"
#include "api/video/video_rotation.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_encoder.h"
#include "common_video/h264/h264_common.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/video_coding/codecs/interface/common_constants.h"
#include "modules/video_coding/include/video_error_codes.h"
#include "rtc_base/checks.h"
#include "rtc_base/time_utils.h"
#include "test/gtest.h"
#include "third_party/libyuv/include/libyuv/compare.h"
#include "third_party/libyuv/include/libyuv/scale.h"
namespace webrtc {
namespace test {
namespace {
const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
const int kMaxBufferedInputFrames = 20;
const VideoEncoder::Capabilities kCapabilities(false);
size_t GetMaxNaluSizeBytes(const EncodedImage& encoded_frame,
const VideoCodecTestFixture::Config& config) {
if (config.codec_settings.codecType != kVideoCodecH264)
return 0;
std::vector<webrtc::H264::NaluIndex> nalu_indices =
webrtc::H264::FindNaluIndices(encoded_frame);
RTC_CHECK(!nalu_indices.empty());
size_t max_size = 0;
for (const webrtc::H264::NaluIndex& index : nalu_indices)
max_size = std::max(max_size, index.payload_size);
return max_size;
}
size_t GetTemporalLayerIndex(const CodecSpecificInfo& codec_specific) {
size_t temporal_idx = 0;
if (codec_specific.codecType == kVideoCodecVP8) {
temporal_idx = codec_specific.codecSpecific.VP8.temporalIdx;
} else if (codec_specific.codecType == kVideoCodecVP9) {
temporal_idx = codec_specific.codecSpecific.VP9.temporal_idx;
}
if (temporal_idx == kNoTemporalIdx) {
temporal_idx = 0;
}
return temporal_idx;
}
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 CalculateFrameQuality(const I420BufferInterface& ref_buffer,
const I420BufferInterface& dec_buffer,
VideoCodecTestStats::FrameStatistics* frame_stat,
bool calc_ssim) {
if (ref_buffer.width() != dec_buffer.width() ||
ref_buffer.height() != dec_buffer.height()) {
RTC_CHECK_GE(ref_buffer.width(), dec_buffer.width());
RTC_CHECK_GE(ref_buffer.height(), dec_buffer.height());
// Downscale reference frame.
rtc::scoped_refptr<I420Buffer> scaled_buffer =
I420Buffer::Create(dec_buffer.width(), dec_buffer.height());
I420Scale(ref_buffer.DataY(), ref_buffer.StrideY(), ref_buffer.DataU(),
ref_buffer.StrideU(), ref_buffer.DataV(), ref_buffer.StrideV(),
ref_buffer.width(), ref_buffer.height(),
scaled_buffer->MutableDataY(), scaled_buffer->StrideY(),
scaled_buffer->MutableDataU(), scaled_buffer->StrideU(),
scaled_buffer->MutableDataV(), scaled_buffer->StrideV(),
scaled_buffer->width(), scaled_buffer->height(),
libyuv::kFilterBox);
CalculateFrameQuality(*scaled_buffer, dec_buffer, frame_stat, calc_ssim);
} else {
const uint64_t sse_y = libyuv::ComputeSumSquareErrorPlane(
dec_buffer.DataY(), dec_buffer.StrideY(), ref_buffer.DataY(),
ref_buffer.StrideY(), dec_buffer.width(), dec_buffer.height());
const uint64_t sse_u = libyuv::ComputeSumSquareErrorPlane(
dec_buffer.DataU(), dec_buffer.StrideU(), ref_buffer.DataU(),
ref_buffer.StrideU(), dec_buffer.width() / 2, dec_buffer.height() / 2);
const uint64_t sse_v = libyuv::ComputeSumSquareErrorPlane(
dec_buffer.DataV(), dec_buffer.StrideV(), ref_buffer.DataV(),
ref_buffer.StrideV(), dec_buffer.width() / 2, dec_buffer.height() / 2);
const size_t num_y_samples = dec_buffer.width() * dec_buffer.height();
const size_t num_u_samples =
dec_buffer.width() / 2 * dec_buffer.height() / 2;
frame_stat->psnr_y = libyuv::SumSquareErrorToPsnr(sse_y, num_y_samples);
frame_stat->psnr_u = libyuv::SumSquareErrorToPsnr(sse_u, num_u_samples);
frame_stat->psnr_v = libyuv::SumSquareErrorToPsnr(sse_v, num_u_samples);
frame_stat->psnr = libyuv::SumSquareErrorToPsnr(
sse_y + sse_u + sse_v, num_y_samples + 2 * num_u_samples);
if (calc_ssim) {
frame_stat->ssim = I420SSIM(ref_buffer, dec_buffer);
}
}
}
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
} // namespace
VideoProcessor::VideoProcessor(const Environment& env,
webrtc::VideoEncoder* encoder,
VideoDecoderList* decoders,
FrameReader* input_frame_reader,
const VideoCodecTestFixture::Config& config,
VideoCodecTestStatsImpl* stats,
IvfFileWriterMap* encoded_frame_writers,
FrameWriterList* decoded_frame_writers)
: config_(config),
num_simulcast_or_spatial_layers_(
std::max(config_.NumberOfSimulcastStreams(),
config_.NumberOfSpatialLayers())),
analyze_frame_quality_(!config_.measure_cpu),
stats_(stats),
encoder_(encoder),
decoders_(decoders),
bitrate_allocator_(CreateBuiltinVideoBitrateAllocatorFactory()->Create(
env,
config_.codec_settings)),
encode_callback_(this),
input_frame_reader_(input_frame_reader),
merged_encoded_frames_(num_simulcast_or_spatial_layers_),
encoded_frame_writers_(encoded_frame_writers),
decoded_frame_writers_(decoded_frame_writers),
last_inputed_frame_num_(0),
last_inputed_timestamp_(0),
first_encoded_frame_(num_simulcast_or_spatial_layers_, true),
last_encoded_frame_num_(num_simulcast_or_spatial_layers_),
first_decoded_frame_(num_simulcast_or_spatial_layers_, true),
last_decoded_frame_num_(num_simulcast_or_spatial_layers_),
last_decoded_frame_buffer_(num_simulcast_or_spatial_layers_),
post_encode_time_ns_(0),
is_finalized_(false) {
// Sanity checks.
Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" This reverts commit 42d8c93ec351b68554825b58a3dc6525a7dc84da. Reason for revert: Got Aliby for FEC test flakes Original change's description: > Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" > > This reverts commit 304e9d2df347630d71fd4423f5971f30dac73e41. > > Reason for revert: Breaks downstream projects. > Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky. > > Original change's description: > > Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current > > > > Bug: webrtc:10191 > > Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27035} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org > > Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10191 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484 > Reviewed-by: Yves Gerey <yvesg@webrtc.org> > Commit-Queue: Yves Gerey <yvesg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27041} TBR=danilchap@webrtc.org,kwiberg@webrtc.org,yvesg@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10191 Change-Id: Id87a17ae415142b8e0b11ba03ae7bad84a473fb0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126720 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Yves Gerey <yvesg@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27056}
2019-03-11 10:28:05 +00:00
RTC_CHECK(TaskQueueBase::Current())
<< "VideoProcessor must be run on a task queue.";
RTC_CHECK(stats_);
RTC_CHECK(encoder_);
RTC_CHECK(decoders_);
RTC_CHECK_EQ(decoders_->size(), num_simulcast_or_spatial_layers_);
RTC_CHECK(input_frame_reader_);
RTC_CHECK(encoded_frame_writers_);
RTC_CHECK(!decoded_frame_writers ||
decoded_frame_writers->size() == num_simulcast_or_spatial_layers_);
// Setup required callbacks for the encoder and decoder and initialize them.
RTC_CHECK_EQ(encoder_->RegisterEncodeCompleteCallback(&encode_callback_),
WEBRTC_VIDEO_CODEC_OK);
// Initialize codecs so that they are ready to receive frames.
RTC_CHECK_EQ(encoder_->InitEncode(
&config_.codec_settings,
VideoEncoder::Settings(
kCapabilities, static_cast<int>(config_.NumberOfCores()),
config_.max_payload_size_bytes)),
WEBRTC_VIDEO_CODEC_OK);
for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
decode_callback_.push_back(
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<VideoProcessorDecodeCompleteCallback>(this, i));
Reland "Update remaining usage of VideoDecoder::InitDecode to Configure" This reverts commit d6da4c23ccda5733f4d8bad3268b539d0c9fc3b7. Reason for revert: downstream project adjusted Original change's description: > Revert "Update remaining usage of VideoDecoder::InitDecode to Configure" > > This reverts commit ca0a08ab600c8d7d00b94492122946ad837b1ef7. > > Reason for revert: Breaks downstream project. > > Original change's description: > > Update remaining usage of VideoDecoder::InitDecode to Configure > > > > Bug: webrtc:13045 > > Change-Id: I5253fddfd613cf0228fc3cd861b91e56558dd34a > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228947 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#34777} > > TBR=danilchap@webrtc.org,sprang@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I1868700a43b5aa4b37e9bcba5af233d24526c974 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:13045 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229024 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34780} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:13045 Change-Id: I5a44e7126f9f2e405f3be6b84698de53b23203a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229183 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34795}
2021-08-18 13:34:36 +00:00
VideoDecoder::Settings decoder_settings;
decoder_settings.set_max_render_resolution(
{config_.codec_settings.width, config_.codec_settings.height});
decoder_settings.set_codec_type(config_.codec_settings.codecType);
decoder_settings.set_number_of_cores(config_.NumberOfCores());
RTC_CHECK(decoders_->at(i)->Configure(decoder_settings));
RTC_CHECK_EQ(decoders_->at(i)->RegisterDecodeCompleteCallback(
decode_callback_.at(i).get()),
WEBRTC_VIDEO_CODEC_OK);
}
}
VideoProcessor::~VideoProcessor() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (!is_finalized_) {
Finalize();
}
// Explicitly reset codecs, in case they don't do that themselves when they
// go out of scope.
RTC_CHECK_EQ(encoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
encoder_->RegisterEncodeCompleteCallback(nullptr);
for (auto& decoder : *decoders_) {
RTC_CHECK_EQ(decoder->Release(), WEBRTC_VIDEO_CODEC_OK);
decoder->RegisterDecodeCompleteCallback(nullptr);
}
// Sanity check.
RTC_CHECK_LE(input_frames_.size(), kMaxBufferedInputFrames);
}
void VideoProcessor::ProcessFrame() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RTC_DCHECK(!is_finalized_);
RTC_DCHECK_GT(target_rates_.size(), 0u);
RTC_DCHECK_EQ(target_rates_.begin()->first, 0u);
RateProfile target_rate =
std::prev(target_rates_.upper_bound(last_inputed_frame_num_))->second;
const size_t frame_number = last_inputed_frame_num_++;
// Get input frame and store for future quality calculation.
Resolution resolution = Resolution({.width = config_.codec_settings.width,
.height = config_.codec_settings.height});
FrameReader::Ratio framerate_scale = FrameReader::Ratio(
{.num = config_.clip_fps.value_or(config_.codec_settings.maxFramerate),
.den = static_cast<int>(config_.codec_settings.maxFramerate)});
rtc::scoped_refptr<I420BufferInterface> buffer =
input_frame_reader_->PullFrame(
/*frame_num*/ nullptr, resolution, framerate_scale);
RTC_CHECK(buffer) << "Tried to read too many frames from the file.";
const size_t timestamp =
last_inputed_timestamp_ +
static_cast<size_t>(kVideoPayloadTypeFrequency / target_rate.input_fps);
VideoFrame input_frame =
VideoFrame::Builder()
.set_video_frame_buffer(buffer)
.set_rtp_timestamp(static_cast<uint32_t>(timestamp))
.set_timestamp_ms(static_cast<int64_t>(timestamp / kMsToRtpTimestamp))
.set_rotation(webrtc::kVideoRotation_0)
.build();
// Store input frame as a reference for quality calculations.
if (config_.decode && !config_.measure_cpu) {
if (input_frames_.size() == kMaxBufferedInputFrames) {
input_frames_.erase(input_frames_.begin());
}
if (config_.reference_width != -1 && config_.reference_height != -1 &&
(input_frame.width() != config_.reference_width ||
input_frame.height() != config_.reference_height)) {
rtc::scoped_refptr<I420Buffer> scaled_buffer = I420Buffer::Create(
config_.codec_settings.width, config_.codec_settings.height);
scaled_buffer->ScaleFrom(*input_frame.video_frame_buffer()->ToI420());
VideoFrame scaled_reference_frame = input_frame;
scaled_reference_frame.set_video_frame_buffer(scaled_buffer);
input_frames_.emplace(frame_number, scaled_reference_frame);
if (config_.reference_width == config_.codec_settings.width &&
config_.reference_height == config_.codec_settings.height) {
// Both encoding and comparison uses the same down-scale factor, reuse
// it for encoder below.
input_frame = scaled_reference_frame;
}
} else {
input_frames_.emplace(frame_number, input_frame);
}
}
last_inputed_timestamp_ = timestamp;
post_encode_time_ns_ = 0;
// Create frame statistics object for all simulcast/spatial layers.
for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
FrameStatistics frame_stat(frame_number, timestamp, i);
stats_->AddFrame(frame_stat);
}
// For the highest measurement accuracy of the encode time, the start/stop
// time recordings should wrap the Encode call as tightly as possible.
const int64_t encode_start_ns = rtc::TimeNanos();
for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
FrameStatistics* frame_stat = stats_->GetFrame(frame_number, i);
frame_stat->encode_start_ns = encode_start_ns;
}
if (input_frame.width() != config_.codec_settings.width ||
input_frame.height() != config_.codec_settings.height) {
rtc::scoped_refptr<I420Buffer> scaled_buffer = I420Buffer::Create(
config_.codec_settings.width, config_.codec_settings.height);
scaled_buffer->ScaleFrom(*input_frame.video_frame_buffer()->ToI420());
input_frame.set_video_frame_buffer(scaled_buffer);
}
// Encode.
const std::vector<VideoFrameType> frame_types =
(frame_number == 0)
? std::vector<VideoFrameType>(num_simulcast_or_spatial_layers_,
VideoFrameType::kVideoFrameKey)
: std::vector<VideoFrameType>(num_simulcast_or_spatial_layers_,
VideoFrameType::kVideoFrameDelta);
const int encode_return_code = encoder_->Encode(input_frame, &frame_types);
for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
FrameStatistics* frame_stat = stats_->GetFrame(frame_number, i);
frame_stat->encode_return_code = encode_return_code;
}
}
void VideoProcessor::SetRates(size_t bitrate_kbps, double framerate_fps) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RTC_DCHECK(!is_finalized_);
target_rates_[last_inputed_frame_num_] =
RateProfile({.target_kbps = bitrate_kbps, .input_fps = framerate_fps});
auto bitrate_allocation =
bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
static_cast<uint32_t>(bitrate_kbps * 1000), framerate_fps));
encoder_->SetRates(
VideoEncoder::RateControlParameters(bitrate_allocation, framerate_fps));
}
int32_t VideoProcessor::VideoProcessorDecodeCompleteCallback::Decoded(
VideoFrame& image) {
// Post the callback to the right task queue, if needed.
if (!task_queue_->IsCurrent()) {
// There might be a limited amount of output buffers, make a copy to make
// sure we don't block the decoder.
VideoFrame copy = VideoFrame::Builder()
.set_video_frame_buffer(I420Buffer::Copy(
*image.video_frame_buffer()->ToI420()))
.set_rotation(image.rotation())
.set_timestamp_us(image.timestamp_us())
.set_id(image.id())
.build();
copy.set_rtp_timestamp(image.rtp_timestamp());
task_queue_->PostTask([this, copy]() {
video_processor_->FrameDecoded(copy, simulcast_svc_idx_);
});
return 0;
}
video_processor_->FrameDecoded(image, simulcast_svc_idx_);
return 0;
}
void VideoProcessor::FrameEncoded(
const webrtc::EncodedImage& encoded_image,
const webrtc::CodecSpecificInfo& codec_specific) {
RTC_DCHECK_RUN_ON(&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.
const int64_t encode_stop_ns = rtc::TimeNanos();
const VideoCodecType codec_type = codec_specific.codecType;
if (config_.encoded_frame_checker) {
config_.encoded_frame_checker->CheckEncodedFrame(codec_type, encoded_image);
}
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
// Layer metadata.
// We could either have simulcast layers or spatial layers.
// TODO(https://crbug.com/webrtc/14891): If we want to support a mix of
// simulcast and SVC we'll also need to consider the case where we have both
// simulcast and spatial indices.
size_t stream_idx = encoded_image.SpatialIndex().value_or(
encoded_image.SimulcastIndex().value_or(0));
size_t temporal_idx = GetTemporalLayerIndex(codec_specific);
FrameStatistics* frame_stat =
stats_->GetFrameWithTimestamp(encoded_image.RtpTimestamp(), stream_idx);
const size_t frame_number = frame_stat->frame_number;
// Ensure that the encode order is monotonically increasing, within this
// simulcast/spatial layer.
RTC_CHECK(first_encoded_frame_[stream_idx] ||
last_encoded_frame_num_[stream_idx] < frame_number);
// Ensure SVC spatial layers are delivered in ascending order.
const size_t num_spatial_layers = config_.NumberOfSpatialLayers();
if (!first_encoded_frame_[stream_idx] && num_spatial_layers > 1) {
for (size_t i = 0; i < stream_idx; ++i) {
RTC_CHECK_LE(last_encoded_frame_num_[i], frame_number);
}
for (size_t i = stream_idx + 1; i < num_simulcast_or_spatial_layers_; ++i) {
RTC_CHECK_GT(frame_number, last_encoded_frame_num_[i]);
}
}
first_encoded_frame_[stream_idx] = false;
last_encoded_frame_num_[stream_idx] = frame_number;
RateProfile target_rate =
std::prev(target_rates_.upper_bound(frame_number))->second;
auto bitrate_allocation =
bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
static_cast<uint32_t>(target_rate.target_kbps * 1000),
target_rate.input_fps));
// Update frame statistics.
frame_stat->encoding_successful = true;
frame_stat->encode_time_us = GetElapsedTimeMicroseconds(
frame_stat->encode_start_ns, encode_stop_ns - post_encode_time_ns_);
frame_stat->target_bitrate_kbps =
bitrate_allocation.GetTemporalLayerSum(stream_idx, temporal_idx) / 1000;
frame_stat->target_framerate_fps = target_rate.input_fps;
frame_stat->length_bytes = encoded_image.size();
frame_stat->frame_type = encoded_image._frameType;
frame_stat->temporal_idx = temporal_idx;
frame_stat->max_nalu_size_bytes = GetMaxNaluSizeBytes(encoded_image, config_);
frame_stat->qp = encoded_image.qp_;
if (codec_type == kVideoCodecVP9) {
const CodecSpecificInfoVP9& vp9_info = codec_specific.codecSpecific.VP9;
frame_stat->inter_layer_predicted = vp9_info.inter_layer_predicted;
frame_stat->non_ref_for_inter_layer_pred =
vp9_info.non_ref_for_inter_layer_pred;
} else {
frame_stat->inter_layer_predicted = false;
frame_stat->non_ref_for_inter_layer_pred = true;
}
const webrtc::EncodedImage* encoded_image_for_decode = &encoded_image;
if (config_.decode || !encoded_frame_writers_->empty()) {
if (num_spatial_layers > 1) {
encoded_image_for_decode = BuildAndStoreSuperframe(
encoded_image, codec_type, frame_number, stream_idx,
frame_stat->inter_layer_predicted);
}
}
if (config_.decode) {
DecodeFrame(*encoded_image_for_decode, stream_idx);
if (codec_specific.end_of_picture && num_spatial_layers > 1) {
// If inter-layer prediction is enabled and upper layer was dropped then
// base layer should be passed to upper layer decoder. Otherwise decoder
// won't be able to decode next superframe.
const EncodedImage* base_image = nullptr;
const FrameStatistics* base_stat = nullptr;
for (size_t i = 0; i < num_spatial_layers; ++i) {
const bool layer_dropped = (first_decoded_frame_[i] ||
last_decoded_frame_num_[i] < frame_number);
// Ensure current layer was decoded.
RTC_CHECK(layer_dropped == false || i != stream_idx);
if (!layer_dropped) {
base_image = &merged_encoded_frames_[i];
base_stat =
stats_->GetFrameWithTimestamp(encoded_image.RtpTimestamp(), i);
} else if (base_image && !base_stat->non_ref_for_inter_layer_pred) {
DecodeFrame(*base_image, i);
}
}
}
} else {
frame_stat->decode_return_code = WEBRTC_VIDEO_CODEC_NO_OUTPUT;
}
// Since frames in higher TLs typically depend on frames in lower TLs,
// write out frames in lower TLs to bitstream dumps of higher TLs.
for (size_t write_temporal_idx = temporal_idx;
write_temporal_idx < config_.NumberOfTemporalLayers();
++write_temporal_idx) {
const VideoProcessor::LayerKey layer_key(stream_idx, write_temporal_idx);
auto it = encoded_frame_writers_->find(layer_key);
if (it != encoded_frame_writers_->cend()) {
RTC_CHECK(it->second->WriteFrame(*encoded_image_for_decode,
config_.codec_settings.codecType));
}
}
if (!config_.encode_in_real_time) {
// To get pure encode time for next layers, measure time spent in encode
// callback and subtract it from encode time of next layers.
post_encode_time_ns_ += rtc::TimeNanos() - encode_stop_ns;
}
}
void VideoProcessor::CalcFrameQuality(const I420BufferInterface& decoded_frame,
FrameStatistics* frame_stat) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
const auto reference_frame = input_frames_.find(frame_stat->frame_number);
RTC_CHECK(reference_frame != input_frames_.cend())
<< "The codecs are either buffering too much, dropping too much, or "
"being too slow relative to the input frame rate.";
// SSIM calculation is not optimized. Skip it in real-time mode.
const bool calc_ssim = !config_.encode_in_real_time;
CalculateFrameQuality(*reference_frame->second.video_frame_buffer()->ToI420(),
decoded_frame, frame_stat, calc_ssim);
frame_stat->quality_analysis_successful = true;
}
void VideoProcessor::WriteDecodedFrame(const I420BufferInterface& decoded_frame,
FrameWriter& frame_writer) {
int input_video_width = config_.codec_settings.width;
int input_video_height = config_.codec_settings.height;
rtc::scoped_refptr<I420Buffer> scaled_buffer;
const I420BufferInterface* scaled_frame;
if (decoded_frame.width() == input_video_width &&
decoded_frame.height() == input_video_height) {
scaled_frame = &decoded_frame;
} else {
EXPECT_DOUBLE_EQ(
static_cast<double>(input_video_width) / input_video_height,
static_cast<double>(decoded_frame.width()) / decoded_frame.height());
scaled_buffer = I420Buffer::Create(input_video_width, input_video_height);
scaled_buffer->ScaleFrom(decoded_frame);
scaled_frame = scaled_buffer.get();
}
// Ensure there is no padding.
RTC_CHECK_EQ(scaled_frame->StrideY(), input_video_width);
RTC_CHECK_EQ(scaled_frame->StrideU(), input_video_width / 2);
RTC_CHECK_EQ(scaled_frame->StrideV(), input_video_width / 2);
RTC_CHECK_EQ(3 * input_video_width * input_video_height / 2,
frame_writer.FrameLength());
RTC_CHECK(frame_writer.WriteFrame(scaled_frame->DataY()));
}
void VideoProcessor::FrameDecoded(const VideoFrame& decoded_frame,
size_t spatial_idx) {
RTC_DCHECK_RUN_ON(&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.
const int64_t decode_stop_ns = rtc::TimeNanos();
FrameStatistics* frame_stat =
stats_->GetFrameWithTimestamp(decoded_frame.rtp_timestamp(), spatial_idx);
const size_t frame_number = frame_stat->frame_number;
if (!first_decoded_frame_[spatial_idx]) {
for (size_t dropped_frame_number = last_decoded_frame_num_[spatial_idx] + 1;
dropped_frame_number < frame_number; ++dropped_frame_number) {
FrameStatistics* dropped_frame_stat =
stats_->GetFrame(dropped_frame_number, spatial_idx);
if (analyze_frame_quality_ && config_.analyze_quality_of_dropped_frames) {
// Calculate frame quality comparing input frame with last decoded one.
CalcFrameQuality(*last_decoded_frame_buffer_[spatial_idx],
dropped_frame_stat);
}
if (decoded_frame_writers_ != nullptr) {
// Fill drops with last decoded frame to make them look like freeze at
// playback and to keep decoded layers in sync.
WriteDecodedFrame(*last_decoded_frame_buffer_[spatial_idx],
*decoded_frame_writers_->at(spatial_idx));
}
}
}
// Ensure that the decode order is monotonically increasing, within this
// simulcast/spatial layer.
RTC_CHECK(first_decoded_frame_[spatial_idx] ||
last_decoded_frame_num_[spatial_idx] < frame_number);
first_decoded_frame_[spatial_idx] = false;
last_decoded_frame_num_[spatial_idx] = frame_number;
// Update frame statistics.
frame_stat->decoding_successful = true;
frame_stat->decode_time_us =
GetElapsedTimeMicroseconds(frame_stat->decode_start_ns, decode_stop_ns);
frame_stat->decoded_width = decoded_frame.width();
frame_stat->decoded_height = decoded_frame.height();
// Skip quality metrics calculation to not affect CPU usage.
if (analyze_frame_quality_ || decoded_frame_writers_) {
// Save last decoded frame to handle possible future drops.
rtc::scoped_refptr<I420BufferInterface> i420buffer =
decoded_frame.video_frame_buffer()->ToI420();
// Copy decoded frame to a buffer without padding/stride such that we can
// dump Y, U and V planes into a file in one shot.
last_decoded_frame_buffer_[spatial_idx] = I420Buffer::Copy(
i420buffer->width(), i420buffer->height(), i420buffer->DataY(),
i420buffer->StrideY(), i420buffer->DataU(), i420buffer->StrideU(),
i420buffer->DataV(), i420buffer->StrideV());
}
if (analyze_frame_quality_) {
CalcFrameQuality(*decoded_frame.video_frame_buffer()->ToI420(), frame_stat);
}
if (decoded_frame_writers_ != nullptr) {
WriteDecodedFrame(*last_decoded_frame_buffer_[spatial_idx],
*decoded_frame_writers_->at(spatial_idx));
}
// Erase all buffered input frames that we have moved past for all
// simulcast/spatial layers. Never buffer more than
// `kMaxBufferedInputFrames` frames, to protect against long runs of
// consecutive frame drops for a particular layer.
const auto min_last_decoded_frame_num = std::min_element(
last_decoded_frame_num_.cbegin(), last_decoded_frame_num_.cend());
const size_t min_buffered_frame_num =
std::max(0, static_cast<int>(frame_number) - kMaxBufferedInputFrames + 1);
RTC_CHECK(min_last_decoded_frame_num != last_decoded_frame_num_.cend());
const auto input_frames_erase_before = input_frames_.lower_bound(
std::max(*min_last_decoded_frame_num, min_buffered_frame_num));
input_frames_.erase(input_frames_.cbegin(), input_frames_erase_before);
}
void VideoProcessor::DecodeFrame(const EncodedImage& encoded_image,
size_t spatial_idx) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
FrameStatistics* frame_stat =
stats_->GetFrameWithTimestamp(encoded_image.RtpTimestamp(), spatial_idx);
frame_stat->decode_start_ns = rtc::TimeNanos();
frame_stat->decode_return_code =
decoders_->at(spatial_idx)->Decode(encoded_image, 0);
}
const webrtc::EncodedImage* VideoProcessor::BuildAndStoreSuperframe(
const EncodedImage& encoded_image,
const VideoCodecType /* codec */,
size_t /* frame_number */,
size_t spatial_idx,
bool inter_layer_predicted) {
// Should only be called for SVC.
RTC_CHECK_GT(config_.NumberOfSpatialLayers(), 1);
EncodedImage base_image;
RTC_CHECK_EQ(base_image.size(), 0);
// Each SVC layer is decoded with dedicated decoder. Find the nearest
// non-dropped base frame and merge it and current frame into superframe.
if (inter_layer_predicted) {
for (int base_idx = static_cast<int>(spatial_idx) - 1; base_idx >= 0;
--base_idx) {
EncodedImage lower_layer = merged_encoded_frames_.at(base_idx);
if (lower_layer.RtpTimestamp() == encoded_image.RtpTimestamp()) {
base_image = lower_layer;
break;
}
}
}
const size_t payload_size_bytes = base_image.size() + encoded_image.size();
auto buffer = EncodedImageBuffer::Create(payload_size_bytes);
if (base_image.size()) {
RTC_CHECK(base_image.data());
memcpy(buffer->data(), base_image.data(), base_image.size());
}
memcpy(buffer->data() + base_image.size(), encoded_image.data(),
encoded_image.size());
EncodedImage copied_image = encoded_image;
copied_image.SetEncodedData(buffer);
if (base_image.size())
copied_image._frameType = base_image._frameType;
// Replace previous EncodedImage for this spatial layer.
merged_encoded_frames_.at(spatial_idx) = std::move(copied_image);
return &merged_encoded_frames_.at(spatial_idx);
}
void VideoProcessor::Finalize() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RTC_DCHECK(!is_finalized_);
is_finalized_ = true;
if (!(analyze_frame_quality_ && config_.analyze_quality_of_dropped_frames) &&
decoded_frame_writers_ == nullptr) {
return;
}
for (size_t spatial_idx = 0; spatial_idx < num_simulcast_or_spatial_layers_;
++spatial_idx) {
if (first_decoded_frame_[spatial_idx]) {
continue; // No decoded frames on this spatial layer.
}
for (size_t dropped_frame_number = last_decoded_frame_num_[spatial_idx] + 1;
dropped_frame_number < last_inputed_frame_num_;
++dropped_frame_number) {
FrameStatistics* frame_stat =
stats_->GetFrame(dropped_frame_number, spatial_idx);
RTC_DCHECK(!frame_stat->decoding_successful);
if (analyze_frame_quality_ && config_.analyze_quality_of_dropped_frames) {
CalcFrameQuality(*last_decoded_frame_buffer_[spatial_idx], frame_stat);
}
if (decoded_frame_writers_ != nullptr) {
WriteDecodedFrame(*last_decoded_frame_buffer_[spatial_idx],
*decoded_frame_writers_->at(spatial_idx));
}
}
}
}
} // namespace test
} // namespace webrtc