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