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https://github.com/mollyim/webrtc.git
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cfea2182f8
Bug: webrtc:12338 Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34696}
117 lines
4.1 KiB
C++
117 lines
4.1 KiB
C++
/*
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* Copyright (c) 2015 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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*/
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#ifndef MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
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#define MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
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// Everything declared in this header is only required when WebRTC is
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// build with H264 support, please do not move anything out of the
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// #ifdef unless needed and tested.
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#ifdef WEBRTC_USE_H264
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#if defined(WEBRTC_WIN) && !defined(__clang__)
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#error "See: bugs.webrtc.org/9213#c13."
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#endif
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#include <memory>
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#include "modules/video_coding/codecs/h264/include/h264.h"
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// CAVEAT: According to ffmpeg docs for avcodec_send_packet, ffmpeg requires a
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// few extra padding bytes after the end of input. And in addition, docs for
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// AV_INPUT_BUFFER_PADDING_SIZE says "If the first 23 bits of the additional
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// bytes are not 0, then damaged MPEG bitstreams could cause overread and
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// segfault."
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//
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// WebRTC doesn't ensure any such padding, and REQUIRES ffmpeg to be compiled
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// with CONFIG_SAFE_BITSTREAM_READER, which is intended to eliminate
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// out-of-bounds reads. ffmpeg docs doesn't say explicitly what effects this
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// flag has on the h.264 decoder or avcodec_send_packet, though, so this is in
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// some way depending on undocumented behavior. If any problems turn up, we may
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// have to add an extra copy operation, to enforce padding before buffers are
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// passed to ffmpeg.
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extern "C" {
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#include "third_party/ffmpeg/libavcodec/avcodec.h"
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} // extern "C"
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#include "common_video/h264/h264_bitstream_parser.h"
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#include "common_video/include/video_frame_buffer_pool.h"
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namespace webrtc {
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struct AVCodecContextDeleter {
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void operator()(AVCodecContext* ptr) const { avcodec_free_context(&ptr); }
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};
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struct AVFrameDeleter {
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void operator()(AVFrame* ptr) const { av_frame_free(&ptr); }
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};
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class H264DecoderImpl : public H264Decoder {
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public:
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H264DecoderImpl();
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~H264DecoderImpl() override;
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// If `codec_settings` is NULL it is ignored. If it is not NULL,
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// `codec_settings->codecType` must be `kVideoCodecH264`.
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int32_t InitDecode(const VideoCodec* codec_settings,
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int32_t number_of_cores) override;
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int32_t Release() override;
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int32_t RegisterDecodeCompleteCallback(
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DecodedImageCallback* callback) override;
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// `missing_frames`, `fragmentation` and `render_time_ms` are ignored.
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int32_t Decode(const EncodedImage& input_image,
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bool /*missing_frames*/,
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int64_t render_time_ms = -1) override;
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const char* ImplementationName() const override;
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private:
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// Called by FFmpeg when it needs a frame buffer to store decoded frames in.
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// The `VideoFrame` returned by FFmpeg at `Decode` originate from here. Their
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// buffers are reference counted and freed by FFmpeg using `AVFreeBuffer2`.
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static int AVGetBuffer2(AVCodecContext* context,
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AVFrame* av_frame,
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int flags);
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// Called by FFmpeg when it is done with a video frame, see `AVGetBuffer2`.
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static void AVFreeBuffer2(void* opaque, uint8_t* data);
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bool IsInitialized() const;
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// Reports statistics with histograms.
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void ReportInit();
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void ReportError();
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// Used by ffmpeg via `AVGetBuffer2()` to allocate I420 images.
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VideoFrameBufferPool ffmpeg_buffer_pool_;
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// Used to allocate NV12 images if NV12 output is preferred.
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VideoFrameBufferPool output_buffer_pool_;
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std::unique_ptr<AVCodecContext, AVCodecContextDeleter> av_context_;
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std::unique_ptr<AVFrame, AVFrameDeleter> av_frame_;
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DecodedImageCallback* decoded_image_callback_;
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bool has_reported_init_;
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bool has_reported_error_;
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webrtc::H264BitstreamParser h264_bitstream_parser_;
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// Decoder should produce this format if possible.
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const VideoFrameBuffer::Type preferred_output_format_;
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};
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} // namespace webrtc
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#endif // WEBRTC_USE_H264
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#endif // MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
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