Intended to make copy construction and assignment of EncodedImage cheaper, but otherwise not have any effect on users of the class. Bug: webrtc:9378, chromium:931692 Change-Id: I22cf8c05f6ef7b7b5cf7ef08fd0dfc5c61211196 Reviewed-on: https://webrtc-review.googlesource.com/c/123442 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26760}
56 lines
1.7 KiB
C++
56 lines
1.7 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 "api/video/encoded_image.h"
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#include <string.h>
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namespace webrtc {
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// FFmpeg's decoder, used by H264DecoderImpl, requires up to 8 bytes padding due
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// to optimized bitstream readers. See avcodec_decode_video2.
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const size_t EncodedImage::kBufferPaddingBytesH264 = 8;
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size_t EncodedImage::GetBufferPaddingBytes(VideoCodecType codec_type) {
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switch (codec_type) {
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case kVideoCodecH264:
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return kBufferPaddingBytesH264;
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default:
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return 0;
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}
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}
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EncodedImage::EncodedImage() : EncodedImage(nullptr, 0, 0) {}
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EncodedImage::EncodedImage(EncodedImage&&) = default;
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EncodedImage::EncodedImage(const EncodedImage&) = default;
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EncodedImage::EncodedImage(uint8_t* buffer, size_t size, size_t capacity)
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: size_(size), buffer_(buffer), capacity_(capacity) {}
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EncodedImage::~EncodedImage() = default;
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EncodedImage& EncodedImage::operator=(EncodedImage&&) = default;
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EncodedImage& EncodedImage::operator=(const EncodedImage&) = default;
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void EncodedImage::Retain() {
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if (buffer_) {
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encoded_data_.SetData(buffer_, size_);
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buffer_ = nullptr;
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}
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}
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void EncodedImage::SetEncodeTime(int64_t encode_start_ms,
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int64_t encode_finish_ms) {
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timing_.encode_start_ms = encode_start_ms;
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timing_.encode_finish_ms = encode_finish_ms;
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}
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} // namespace webrtc
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