ffmpeg结构体以及函数介绍（三）
1 AVPacket
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typedef struct AVPacket { /** * Presentation timestamp in AVStream->time_base units; the time at which * the decompressed packet will be presented to the user. * Can be AV_NOPTS_VALUE if it is not stored in the file. * pts MUST be larger or equal to dts as presentation cannot happen before * decompression, unless one wants to view hex dumps. Some formats misuse * the terms dts and pts/cts to mean something different. Such timestamps * must be converted to true pts/dts before they are stored in AVPacket. */ int64_t pts; /** * Decompression timestamp in AVStream->time_base units; the time at which * the packet is decompressed. * Can be AV_NOPTS_VALUE if it is not stored in the file. */ int64_t dts; uint8_t *data; int size; int stream_index; int flags; int duration; . . . } AVPacket // AVPacket是个很重要的结构,该结构在读媒体源文件和写输出文件时都需要用到 // int64_t pts; 显示时间戳 // int64_t dts; 解码时间戳 // uint8_t *data; 包数据 // int size; 包数据长度 // int stream_index; 包所属流序号 // int duration; 时长 // 以上信息,如果是在读媒体源文件那么avcodec会初始化,如果是输出文件,用户需要对以上信息赋值
2 av_init_packet()
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/** * Initialize optional fields of a packet with default values. * * @param pkt packet */ void av_init_packet(AVPacket *pkt); // 使用默认值初始化AVPacket // 定义AVPacket对象后,请使用av_init_packet进行初始化
3 av_free_packet()
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/** * Free a packet. * * @param pkt packet to free */ void av_free_packet(AVPacket *pkt); // 释放AVPacket对象
4 av_read_frame()
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/** * Return the next frame of a stream. * This function returns what is stored in the file, and does not validate * that what is there are valid frames for the decoder. It will split what is * stored in the file into frames and return one for each call. It will not * omit invalid data between valid frames so as to give the decoder the maximum * information possible for decoding. * * The returned packet is valid * until the next av_read_frame() or until av_close_input_file() and * must be freed with av_free_packet. For video, the packet contains * exactly one frame. For audio, it contains an integer number of * frames if each frame has a known fixed size (e.g. PCM or ADPCM * data). If the audio frames have a variable size (e.g. MPEG audio), * then it contains one frame. * * pkt->pts, pkt->dts and pkt->duration are always set to correct * values in AVStream.time_base units (and guessed if the format cannot * provide them). pkt->pts can be AV_NOPTS_VALUE if the video format * has B-frames, so it is better to rely on pkt->dts if you do not * decompress the payload. * * @return 0 if OK, < 0 on error or end of file */ int av_read_frame(AVFormatContext *s, AVPacket *pkt); // 从输入源文件容器中读取一个AVPacket数据包 // 该函数读出的包并不每次都是有效的,对于读出的包我们都应该进行相应的解码(视频解码/音频解码), // 在返回值>=0时,循环调用该函数进行读取,循环调用之前请调用av_free_packet函数清理AVPacket
5 avcodec_decode_video2()
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/** * Decode the video frame of size avpkt->size from avpkt->data into picture. * Some decoders may support multiple frames in a single AVPacket, such * decoders would then just decode the first frame. * * @warning The input buffer must be FF_INPUT_BUFFER_PADDING_SIZE larger than * the actual read bytes because some optimized bitstream readers read 32 or 64 * bits at once and could read over the end. * * @warning The end of the input buffer buf should be set to 0 to ensure that * no overreading happens for damaged MPEG streams. * * @note You might have to align the input buffer avpkt->data. * The alignment requirements depend on the CPU: on some CPUs it isn't * necessary at all, on others it won't work at all if not aligned and on others * it will work but it will have an impact on performance. * * In practice, avpkt->data should have 4 byte alignment at minimum. * * @note Some codecs have a delay between input and output, these need to be * fed with avpkt->data=NULL, avpkt->size=0 at the end to return the remaining frames. * * @param avctx the codec context * @param[out] picture The AVFrame in which the decoded video frame will be stored. * Use avcodec_alloc_frame to get an AVFrame, the codec will * allocate memory for the actual bitmap. * with default get/release_buffer(), the decoder frees/reuses the bitmap as it sees fit. * with overridden get/release_buffer() (needs CODEC_CAP_DR1) the user decides into what buffer the decoder * decodes and the decoder tells the user once it does not need the data anymore, * the user app can at this point free/reuse/keep the memory as it sees fit. * * @param[in] avpkt The input AVpacket containing the input buffer. * You can create such packet with av_init_packet() and by then setting * data and size, some decoders might in addition need other fields like * flags&AV_PKT_FLAG_KEY. All decoders are designed to use the least * fields possible. * @param[in,out] got_picture_ptr Zero if no frame could be decompressed, otherwise, it is nonzero. * @return On error a negative value is returned, otherwise the number of bytes * used or zero if no frame could be decompressed. */ int avcodec_decode_video2(AVCodecContext *avctx, AVFrame *picture, int *got_picture_ptr, AVPacket *avpkt); // 解码视频流AVPacket // 使用av_read_frame读取媒体流后需要进行判断,如果为视频流则调用该函数解码 // 返回结果<0时失败,此时程序应该退出检查原因 // 返回>=0时正常,假设 读取包为:AVPacket vPacket 返回值为 int vLen; 每次解码正常时,对vPacket做 // 如下处理: // vPacket.size -= vLen; // vPacket.data += vLen; // 如果 vPacket.size==0,则继续读下一流包,否则继续调度该方法进行解码,直到vPacket.size==0 // 返回 got_picture_ptr > 0 时,表示解码到了AVFrame *picture,其后可以对picture进程处理
6 avcodec_decode_audio3()
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/** * Decode the audio frame of size avpkt->size from avpkt->data into samples. * Some decoders may support multiple frames in a single AVPacket, such * decoders would then just decode the first frame. In this case, * avcodec_decode_audio3 has to be called again with an AVPacket that contains * the remaining data in order to decode the second frame etc. * If no frame * could be outputted, frame_size_ptr is zero. Otherwise, it is the * decompressed frame size in bytes. * * @warning You must set frame_size_ptr to the allocated size of the * output buffer before calling avcodec_decode_audio3(). * * @warning The input buffer must be FF_INPUT_BUFFER_PADDING_SIZE larger than * the actual read bytes because some optimized bitstream readers read 32 or 64 * bits at once and could read over the end. * * @warning The end of the input buffer avpkt->data should be set to 0 to ensure that * no overreading happens for damaged MPEG streams. * * @note You might have to align the input buffer avpkt->data and output buffer * samples. The alignment requirements depend on the CPU: On some CPUs it isn't * necessary at all, on others it won't work at all if not aligned and on others * it will work but it will have an impact on performance. * * In practice, avpkt->data should have 4 byte alignment at minimum and * samples should be 16 byte aligned unless the CPU doesn't need it * (AltiVec and SSE do). * * @param avctx the codec context * @param[out] samples the output buffer, sample type in avctx->sample_fmt * @param[in,out] frame_size_ptr the output buffer size in bytes * @param[in] avpkt The input AVPacket containing the input buffer. * You can create such packet with av_init_packet() and by then setting * data and size, some decoders might in addition need other fields. * All decoders are designed to use the least fields possible though. * @return On error a negative value is returned, otherwise the number of bytes * used or zero if no frame data was decompressed (used) from the input AVPacket. */ int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples, int *frame_size_ptr, AVPacket *avpkt); // 解码音频流AVPacket // 使用av_read_frame读取媒体流后需要进行判断,如果为音频流则调用该函数解码 // 返回结果<0时失败,此时程序应该退出检查原因 // 返回>=0时正常,假设 读取包为:AVPacket vPacket 返回值为 int vLen; 每次解码正常时,对vPacket做 // 如下处理: // vPacket.size -= vLen; // vPacket.data += vLen; // 如果 vPacket.size==0,则继续读下一流包,否则继续调度该方法进行解码,直到vPacket.size==0 //