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317a1f09ed
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}
127 lines
3.8 KiB
C++
127 lines
3.8 KiB
C++
/*
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* Copyright (c) 2013 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 "test/fake_decoder.h"
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#include <string.h>
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#include <memory>
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#include "api/scoped_refptr.h"
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#include "api/video/i420_buffer.h"
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#include "api/video/video_frame.h"
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#include "api/video/video_frame_buffer.h"
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#include "api/video/video_rotation.h"
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#include "modules/video_coding/include/video_error_codes.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/time_utils.h"
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namespace webrtc {
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namespace test {
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namespace {
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const int kDefaultWidth = 320;
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const int kDefaultHeight = 180;
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} // namespace
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FakeDecoder::FakeDecoder() : FakeDecoder(nullptr) {}
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FakeDecoder::FakeDecoder(TaskQueueFactory* task_queue_factory)
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: callback_(nullptr),
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width_(kDefaultWidth),
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height_(kDefaultHeight),
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task_queue_factory_(task_queue_factory),
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decode_delay_ms_(0) {}
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int32_t FakeDecoder::InitDecode(const VideoCodec* config,
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int32_t number_of_cores) {
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t FakeDecoder::Decode(const EncodedImage& input,
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bool missing_frames,
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int64_t render_time_ms) {
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if (input._encodedWidth > 0 && input._encodedHeight > 0) {
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width_ = input._encodedWidth;
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height_ = input._encodedHeight;
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}
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rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(width_, height_);
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I420Buffer::SetBlack(buffer);
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VideoFrame frame = VideoFrame::Builder()
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.set_video_frame_buffer(buffer)
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.set_rotation(webrtc::kVideoRotation_0)
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.set_timestamp_ms(render_time_ms)
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.build();
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frame.set_timestamp(input.Timestamp());
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frame.set_ntp_time_ms(input.ntp_time_ms_);
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if (decode_delay_ms_ == 0 || !task_queue_) {
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callback_->Decoded(frame);
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} else {
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task_queue_->PostDelayedTask(
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[frame, this]() {
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VideoFrame copy = frame;
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callback_->Decoded(copy);
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},
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decode_delay_ms_);
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}
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return WEBRTC_VIDEO_CODEC_OK;
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}
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void FakeDecoder::SetDelayedDecoding(int decode_delay_ms) {
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RTC_CHECK(task_queue_factory_);
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if (!task_queue_) {
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task_queue_ =
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std::make_unique<rtc::TaskQueue>(task_queue_factory_->CreateTaskQueue(
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"fake_decoder", TaskQueueFactory::Priority::NORMAL));
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}
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decode_delay_ms_ = decode_delay_ms;
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}
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int32_t FakeDecoder::RegisterDecodeCompleteCallback(
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DecodedImageCallback* callback) {
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callback_ = callback;
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t FakeDecoder::Release() {
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return WEBRTC_VIDEO_CODEC_OK;
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}
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const char* FakeDecoder::kImplementationName = "fake_decoder";
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const char* FakeDecoder::ImplementationName() const {
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return kImplementationName;
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}
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int32_t FakeH264Decoder::Decode(const EncodedImage& input,
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bool missing_frames,
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int64_t render_time_ms) {
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uint8_t value = 0;
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for (size_t i = 0; i < input.size(); ++i) {
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uint8_t kStartCode[] = {0, 0, 0, 1};
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if (i < input.size() - sizeof(kStartCode) &&
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!memcmp(&input.data()[i], kStartCode, sizeof(kStartCode))) {
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i += sizeof(kStartCode) + 1; // Skip start code and NAL header.
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}
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if (input.data()[i] != value) {
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RTC_CHECK_EQ(value, input.data()[i])
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<< "Bitstream mismatch between sender and receiver.";
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return -1;
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}
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++value;
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}
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return FakeDecoder::Decode(input, missing_frames, render_time_ms);
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}
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} // namespace test
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} // namespace webrtc
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