| /* |
| * Copyright (c) 2015 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. |
| * |
| */ |
| |
| #ifndef MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_ |
| #define MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_ |
| |
| // Everything declared in this header is only required when WebRTC is |
| // build with H264 support, please do not move anything out of the |
| // #ifdef unless needed and tested. |
| #ifdef WEBRTC_USE_H264 |
| |
| #if defined(WEBRTC_WIN) && !defined(__clang__) |
| #error "See: bugs.webrtc.org/9213#c13." |
| #endif |
| |
| #include <memory> |
| |
| #include "modules/video_coding/codecs/h264/include/h264.h" |
| |
| // CAVEAT: According to ffmpeg docs for avcodec_send_packet, ffmpeg requires a |
| // few extra padding bytes after the end of input. And in addition, docs for |
| // AV_INPUT_BUFFER_PADDING_SIZE says "If the first 23 bits of the additional |
| // bytes are not 0, then damaged MPEG bitstreams could cause overread and |
| // segfault." |
| // |
| // WebRTC doesn't ensure any such padding, and REQUIRES ffmpeg to be compiled |
| // with CONFIG_SAFE_BITSTREAM_READER, which is intended to eliminate |
| // out-of-bounds reads. ffmpeg docs doesn't say explicitly what effects this |
| // flag has on the h.264 decoder or avcodec_send_packet, though, so this is in |
| // some way depending on undocumented behavior. If any problems turn up, we may |
| // have to add an extra copy operation, to enforce padding before buffers are |
| // passed to ffmpeg. |
| |
| extern "C" { |
| #include "third_party/ffmpeg/libavcodec/avcodec.h" |
| } // extern "C" |
| |
| #include "common_video/h264/h264_bitstream_parser.h" |
| #include "common_video/include/i420_buffer_pool.h" |
| |
| namespace webrtc { |
| |
| struct AVCodecContextDeleter { |
| void operator()(AVCodecContext* ptr) const { avcodec_free_context(&ptr); } |
| }; |
| struct AVFrameDeleter { |
| void operator()(AVFrame* ptr) const { av_frame_free(&ptr); } |
| }; |
| |
| class H264DecoderImpl : public H264Decoder { |
| public: |
| H264DecoderImpl(); |
| ~H264DecoderImpl() override; |
| |
| // If |codec_settings| is NULL it is ignored. If it is not NULL, |
| // |codec_settings->codecType| must be |kVideoCodecH264|. |
| int32_t InitDecode(const VideoCodec* codec_settings, |
| int32_t number_of_cores) override; |
| int32_t Release() override; |
| |
| int32_t RegisterDecodeCompleteCallback( |
| DecodedImageCallback* callback) override; |
| |
| // |missing_frames|, |fragmentation| and |render_time_ms| are ignored. |
| int32_t Decode(const EncodedImage& input_image, |
| bool /*missing_frames*/, |
| int64_t render_time_ms = -1) override; |
| |
| const char* ImplementationName() const override; |
| |
| private: |
| const bool kEnable8bitHdrFix_; |
| // Called by FFmpeg when it needs a frame buffer to store decoded frames in. |
| // The |VideoFrame| returned by FFmpeg at |Decode| originate from here. Their |
| // buffers are reference counted and freed by FFmpeg using |AVFreeBuffer2|. |
| static int AVGetBuffer2(AVCodecContext* context, |
| AVFrame* av_frame, |
| int flags); |
| // Called by FFmpeg when it is done with a video frame, see |AVGetBuffer2|. |
| static void AVFreeBuffer2(void* opaque, uint8_t* data); |
| |
| bool IsInitialized() const; |
| |
| // Reports statistics with histograms. |
| void ReportInit(); |
| void ReportError(); |
| |
| I420BufferPool pool_; |
| std::unique_ptr<AVCodecContext, AVCodecContextDeleter> av_context_; |
| std::unique_ptr<AVFrame, AVFrameDeleter> av_frame_; |
| |
| DecodedImageCallback* decoded_image_callback_; |
| |
| bool has_reported_init_; |
| bool has_reported_error_; |
| |
| webrtc::H264BitstreamParser h264_bitstream_parser_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_USE_H264 |
| |
| #endif // MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_ |