5.av_write_frame()

av_write_frame()用于输出一帧视音频数据

/** 
 * Write a packet to an output media file. 
 * 
 * This function passes the packet directly to the muxer, without any buffering 
 * or reordering. The caller is responsible for correctly interleaving the 
 * packets if the format requires it. Callers that want libavformat to handle 
 * the interleaving should call av_interleaved_write_frame() instead of this 
 * function. 
 * 
 * @param s media file handle 
 * @param pkt The packet containing the data to be written. Note that unlike 
 *            av_interleaved_write_frame(), this function does not take 
 *            ownership of the packet passed to it (though some muxers may make 
 *            an internal reference to the input packet). 
 *            <br> 
 *            This parameter can be NULL (at any time, not just at the end), in 
 *            order to immediately flush data buffered within the muxer, for 
 *            muxers that buffer up data internally before writing it to the 
 *            output. 
 *            <br> 
 *            Packet's @ref AVPacket.stream_index "stream_index" field must be 
 *            set to the index of the corresponding stream in @ref 
 *            AVFormatContext.streams "s->streams". It is very strongly 
 *            recommended that timing information (@ref AVPacket.pts "pts", @ref 
 *            AVPacket.dts "dts", @ref AVPacket.duration "duration") is set to 
 *            correct values. 
 * @return < 0 on error, = 0 if OK, 1 if flushed and there is no more data to flush 
 * 
 * @see av_interleaved_write_frame() 
 */  
int av_write_frame(AVFormatContext *s, AVPacket *pkt);

简单解释一下它的参数的含义：

• s：用于输出的AVFormatContext。
• pkt：等待输出的AVPacket。
• 函数正常执行后返回值等于0。

函数调用关系图：

源码：

int av_write_frame(AVFormatContext *s, AVPacket *pkt)  
{  
    int ret;  

    ret = check_packet(s, pkt);  
    if (ret < 0)  
        return ret;  
    //Packet为NULL，Flush Encoder  
    if (!pkt) {  
        if (s->oformat->flags & AVFMT_ALLOW_FLUSH) {  
            ret = s->oformat->write_packet(s, NULL);  
            if (s->flush_packets && s->pb && s->pb->error >= 0 && s->flags & AVFMT_FLAG_FLUSH_PACKETS)  
                avio_flush(s->pb);  
            if (ret >= 0 && s->pb && s->pb->error < 0)  
                ret = s->pb->error;  
            return ret;  
        }  
        return 1;  
    }  

    ret = compute_pkt_fields2(s, s->streams[pkt->stream_index], pkt);  

    if (ret < 0 && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))  
        return ret;  
    //写入  
    ret = write_packet(s, pkt);  
    if (ret >= 0 && s->pb && s->pb->error < 0)  
        ret = s->pb->error;  

    if (ret >= 0)  
        s->streams[pkt->stream_index]->nb_frames++;  
    return ret;  
}

从源代码可以看出，av_write_frame()主要完成了以下几步工作：

• （1）调用check_packet()做一些简单的检测
• （2）调用compute_pkt_fields2()设置AVPacket的一些属性值
• （3）调用write_packet()写入数据

check_packet()

static int check_packet(AVFormatContext *s, AVPacket *pkt)  
{  
    if (!pkt)  
        return 0;  

    if (pkt->stream_index < 0 || pkt->stream_index >= s->nb_streams) {  
        av_log(s, AV_LOG_ERROR, "Invalid packet stream index: %d\n",  
               pkt->stream_index);  
        return AVERROR(EINVAL);  
    }  

    if (s->streams[pkt->stream_index]->codec->codec_type == AVMEDIA_TYPE_ATTACHMENT) {  
        av_log(s, AV_LOG_ERROR, "Received a packet for an attachment stream.\n");  
        return AVERROR(EINVAL);  
    }  

    return 0;  
}

从代码中可以看出，check_packet()的功能比较简单：首先检查一下输入的AVPacket是否为空，如果为空，则是直接返回；然后检查一下AVPacket的stream_index（标记了该AVPacket所属的AVStream）设置是否正常，如果为负数或者大于AVStream的个数，则返回错误信息；最后检查AVPacket所属的AVStream是否属于attachment stream，这个地方没见过，目前还没有研究

compute_pkt_fields2()

//FIXME merge with compute_pkt_fields  
static int compute_pkt_fields2(AVFormatContext *s, AVStream *st, AVPacket *pkt)  
{  
    int delay = FFMAX(st->codec->has_b_frames, st->codec->max_b_frames > 0);  
    int num, den, i;  
    int frame_size;  

    av_dlog(s, "compute_pkt_fields2: pts:%s dts:%s cur_dts:%s b:%d size:%d st:%d\n",  
            av_ts2str(pkt->pts), av_ts2str(pkt->dts), av_ts2str(st->cur_dts), delay, pkt->size, pkt->stream_index);  

    if (pkt->duration < 0 && st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE) {  
        av_log(s, AV_LOG_WARNING, "Packet with invalid duration %d in stream %d\n",  
               pkt->duration, pkt->stream_index);  
        pkt->duration = 0;  
    }  

    /* duration field */  
    if (pkt->duration == 0) {  
        ff_compute_frame_duration(s, &num, &den, st, NULL, pkt);  
        if (den && num) {  
            pkt->duration = av_rescale(1, num * (int64_t)st->time_base.den * st->codec->ticks_per_frame, den * (int64_t)st->time_base.num);  
        }  
    }  

    if (pkt->pts == AV_NOPTS_VALUE && pkt->dts != AV_NOPTS_VALUE && delay == 0)  
        pkt->pts = pkt->dts;  

    //XXX/FIXME this is a temporary hack until all encoders output pts  
    if ((pkt->pts == 0 || pkt->pts == AV_NOPTS_VALUE) && pkt->dts == AV_NOPTS_VALUE && !delay) {  
        static int warned;  
        if (!warned) {  
            av_log(s, AV_LOG_WARNING, "Encoder did not produce proper pts, making some up.\n");  
            warned = 1;  
        }  
        pkt->dts =  
//        pkt->pts= st->cur_dts;  
            pkt->pts = st->pts.val;  
    }  

    //calculate dts from pts  
    if (pkt->pts != AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE && delay <= MAX_REORDER_DELAY) {  
        st->pts_buffer[0] = pkt->pts;  
        for (i = 1; i < delay + 1 && st->pts_buffer[i] == AV_NOPTS_VALUE; i++)  
            st->pts_buffer[i] = pkt->pts + (i - delay - 1) * pkt->duration;  
        for (i = 0; i<delay && st->pts_buffer[i] > st->pts_buffer[i + 1]; i++)  
            FFSWAP(int64_t, st->pts_buffer[i], st->pts_buffer[i + 1]);  

        pkt->dts = st->pts_buffer[0];  
    }  

    if (st->cur_dts && st->cur_dts != AV_NOPTS_VALUE &&  
        ((!(s->oformat->flags & AVFMT_TS_NONSTRICT) &&  
          st->cur_dts >= pkt->dts) || st->cur_dts > pkt->dts)) {  
        av_log(s, AV_LOG_ERROR,  
               "Application provided invalid, non monotonically increasing dts to muxer in stream %d: %s >= %s\n",  
               st->index, av_ts2str(st->cur_dts), av_ts2str(pkt->dts));  
        return AVERROR(EINVAL);  
    }  
    if (pkt->dts != AV_NOPTS_VALUE && pkt->pts != AV_NOPTS_VALUE && pkt->pts < pkt->dts) {  
        av_log(s, AV_LOG_ERROR,  
               "pts (%s) < dts (%s) in stream %d\n",  
               av_ts2str(pkt->pts), av_ts2str(pkt->dts),  
               st->index);  
        return AVERROR(EINVAL);  
    }  

    av_dlog(s, "av_write_frame: pts2:%s dts2:%s\n",  
            av_ts2str(pkt->pts), av_ts2str(pkt->dts));  
    st->cur_dts = pkt->dts;  
    st->pts.val = pkt->dts;  

    /* update pts */  
    switch (st->codec->codec_type) {  
    case AVMEDIA_TYPE_AUDIO:  
        frame_size = (pkt->flags & AV_PKT_FLAG_UNCODED_FRAME) ?  
                     ((AVFrame *)pkt->data)->nb_samples :  
                     av_get_audio_frame_duration(st->codec, pkt->size);  

        /* HACK/FIXME, we skip the initial 0 size packets as they are most 
         * likely equal to the encoder delay, but it would be better if we 
         * had the real timestamps from the encoder */  
        if (frame_size >= 0 && (pkt->size || st->pts.num != st->pts.den >> 1 || st->pts.val)) {  
            frac_add(&st->pts, (int64_t)st->time_base.den * frame_size);  
        }  
        break;  
    case AVMEDIA_TYPE_VIDEO:  
        frac_add(&st->pts, (int64_t)st->time_base.den * st->codec->time_base.num);  
        break;  
    }  
    return 0;  
}

从代码中可以看出，compute_pkt_fields2()主要有两方面的功能：一方面用于计算AVPacket的duration， dts等信息；另一方面用于检查pts、dts这些参数的合理性（例如PTS是否一定大于DTS）

AVOutputFormat->write_packet()

/** 
 * Make timestamps non negative, move side data from payload to internal struct, call muxer, and restore 
 * sidedata. 
 * 
 * FIXME: this function should NEVER get undefined pts/dts beside when the 
 * AVFMT_NOTIMESTAMPS is set. 
 * Those additional safety checks should be dropped once the correct checks 
 * are set in the callers. 
 */  
static int write_packet(AVFormatContext *s, AVPacket *pkt)  
{  
    int ret, did_split;  

    if (s->output_ts_offset) {  
        AVStream *st = s->streams[pkt->stream_index];  
        int64_t offset = av_rescale_q(s->output_ts_offset, AV_TIME_BASE_Q, st->time_base);  

        if (pkt->dts != AV_NOPTS_VALUE)  
            pkt->dts += offset;  
        if (pkt->pts != AV_NOPTS_VALUE)  
            pkt->pts += offset;  
    }  

    if (s->avoid_negative_ts > 0) {  
        AVStream *st = s->streams[pkt->stream_index];  
        int64_t offset = st->mux_ts_offset;  

        if (s->offset == AV_NOPTS_VALUE && pkt->dts != AV_NOPTS_VALUE &&  
            (pkt->dts < 0 || s->avoid_negative_ts == AVFMT_AVOID_NEG_TS_MAKE_ZERO)) {  
            s->offset = -pkt->dts;  
            s->offset_timebase = st->time_base;  
        }  

        if (s->offset != AV_NOPTS_VALUE && !offset) {  
            offset = st->mux_ts_offset =  
                av_rescale_q_rnd(s->offset,  
                                 s->offset_timebase,  
                                 st->time_base,  
                                 AV_ROUND_UP);  
        }  

        if (pkt->dts != AV_NOPTS_VALUE)  
            pkt->dts += offset;  
        if (pkt->pts != AV_NOPTS_VALUE)  
            pkt->pts += offset;  

        av_assert2(pkt->dts == AV_NOPTS_VALUE || pkt->dts >= 0 || s->max_interleave_delta > 0);  
        if (pkt->dts != AV_NOPTS_VALUE && pkt->dts < 0) {  
            av_log(s, AV_LOG_WARNING,  
                   "Packets poorly interleaved, failed to avoid negative "  
                   "timestamp %s in stream %d.\n"  
                   "Try -max_interleave_delta 0 as a possible workaround.\n",  
                   av_ts2str(pkt->dts),  
                   pkt->stream_index  
            );  
        }  
    }  

    did_split = av_packet_split_side_data(pkt);  
    if ((pkt->flags & AV_PKT_FLAG_UNCODED_FRAME)) {  
        AVFrame *frame = (AVFrame *)pkt->data;  
        av_assert0(pkt->size == UNCODED_FRAME_PACKET_SIZE);  
        ret = s->oformat->write_uncoded_frame(s, pkt->stream_index, &frame, 0);  
        av_frame_free(&frame);  
    } else {  
        //写入  
        ret = s->oformat->write_packet(s, pkt);  
    }  

    if (s->flush_packets && s->pb && ret >= 0 && s->flags & AVFMT_FLAG_FLUSH_PACKETS)  
        avio_flush(s->pb);  

    if (did_split)  
        av_packet_merge_side_data(pkt);  

    return ret;  
}

write_packet()函数最关键的地方就是调用了AVOutputFormat中写入数据的方法。如果AVPacket中的flag标记中包含AV_PKT_FLAG_UNCODED_FRAME，就会调用AVOutputFormat的write_uncoded_frame()函数；如果不包含那个标记，就会调用write_packet()函数。write_packet()实际上是一个函数指针，指向特定的AVOutputFormat中的实现函数

例如，我们看一下FLV对应的AVOutputFormat，位于

AVOutputFormat ff_flv_muxer = {  
    .name           = "flv",  
    .long_name      = NULL_IF_CONFIG_SMALL("FLV (Flash Video)"),  
    .mime_type      = "video/x-flv",  
    .extensions     = "flv",  
    .priv_data_size = sizeof(FLVContext),  
    .audio_codec    = CONFIG_LIBMP3LAME ? AV_CODEC_ID_MP3 : AV_CODEC_ID_ADPCM_SWF,  
    .video_codec    = AV_CODEC_ID_FLV1,  
    .write_header   = flv_write_header,  
    .write_packet   = flv_write_packet,  
    .write_trailer  = flv_write_trailer,  
    .codec_tag      = (const AVCodecTag* const []) {  
                          flv_video_codec_ids, flv_audio_codec_ids, 0  
                      },  
    .flags          = AVFMT_GLOBALHEADER | AVFMT_VARIABLE_FPS |  
                      AVFMT_TS_NONSTRICT,  
};

从ff_flv_muxer的定义可以看出，write_packet()指向的是flv_write_packet()函数。在看flv_write_packet()函数的定义之前，我们先回顾一下FLV封装格式的结构。

FLV封装格式

从FLV的封装格式结构可以看出，它的文件数据是一个一个的Tag连接起来的，中间间隔包含着Previous Tag Size。因此，flv_write_packet()函数的任务就是写入一个Tag和Previous Tag Size。下面简单记录一下Tag Data的格式。Tag Data根据Tag的Type不同而不同：可以分为音频Tag Data，视频Tag Data以及Script Tag Data。下面简述一下音频Tag Data和视频Tag Data。

Audio Tag Data

Audio Tag开始的第1个字节包含了音频数据的参数信息，从第2个字节开始为音频流数据。 第1个字节的前4位的数值表示了音频数据格式：

• 0 = Linear PCM, platform endian
• 1 = ADPCM
• 2 = MP3
• 3 = Linear PCM, little endian
• 4 = Nellymoser 16-kHz mono
• 5 = Nellymoser 8-kHz mono
• 6 = Nellymoser
• 7 = G.711 A-law logarithmic PCM
• 8 = G.711 mu-law logarithmic PCM
• 9 = reserved
• 10 = AAC
• 14 = MP3 8-Khz
• 15 = Device-specific sound 第1个字节的第5-6位的数值表示采样率：0 = 5.5kHz，1 = 11KHz，2 = 22 kHz，3 = 44 kHz。 第1个字节的第7位表示采样精度：0 = 8bits，1 = 16bits。 第1个字节的第8位表示音频类型：0 = sndMono，1 = sndStereo。 其中，当音频编码为AAC的时候，第一个字节后面存储的是AACAUDIODATA，格式如下所示。

Video Tag Data

Video Tag也用开始的第1个字节包含视频数据的参数信息，从第2个字节为视频流数据。

• 第1个字节的前4位的数值表示帧类型（FrameType）：
  • 1: keyframe (for AVC, a seekableframe)（关键帧）
  • 2: inter frame (for AVC, a nonseekableframe)
  • 3: disposable inter frame (H.263only)
  • 4: generated keyframe (reservedfor server use only)
  • 5: video info/command frame
• 第1个字节的后4位的数值表示视频编码ID（CodecID）：
  • 1: JPEG (currently unused)
  • 2: Sorenson H.263
  • 3: Screen video
  • 4: On2 VP6
  • 5: On2 VP6 with alpha channel
  • 6: Screen video version 2
  • 7: AVC

其中，当音频编码为AVC（H.264）的时候，第一个字节后面存储的是AVCVIDEOPACKET，格式如下所示。

flv_write_packet()

static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)  
{  
    AVIOContext *pb      = s->pb;  
    AVCodecContext *enc  = s->streams[pkt->stream_index]->codec;  
    FLVContext *flv      = s->priv_data;  
    FLVStreamContext *sc = s->streams[pkt->stream_index]->priv_data;  
    unsigned ts;  
    int size = pkt->size;  
    uint8_t *data = NULL;  
    int flags = -1, flags_size, ret;  

    if (enc->codec_id == AV_CODEC_ID_VP6F || enc->codec_id == AV_CODEC_ID_VP6A ||  
        enc->codec_id == AV_CODEC_ID_VP6  || enc->codec_id == AV_CODEC_ID_AAC)  
        flags_size = 2;  
    else if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4)  
        flags_size = 5;  
    else  
        flags_size = 1;  

    if (flv->delay == AV_NOPTS_VALUE)  
        flv->delay = -pkt->dts;  

    if (pkt->dts < -flv->delay) {  
        av_log(s, AV_LOG_WARNING,  
               "Packets are not in the proper order with respect to DTS\n");  
        return AVERROR(EINVAL);  
    }  

    ts = pkt->dts + flv->delay; // add delay to force positive dts  

    if (s->event_flags & AVSTREAM_EVENT_FLAG_METADATA_UPDATED) {  
        write_metadata(s, ts);  
        s->event_flags &= ~AVSTREAM_EVENT_FLAG_METADATA_UPDATED;  
    }  
    //Tag Header  
    switch (enc->codec_type) {  
    case AVMEDIA_TYPE_VIDEO:  
        //Type  
        avio_w8(pb, FLV_TAG_TYPE_VIDEO);  

        flags = enc->codec_tag;  
        if (flags == 0) {  
            av_log(s, AV_LOG_ERROR,  
                   "Video codec '%s' is not compatible with FLV\n",  
                   avcodec_get_name(enc->codec_id));  
            return AVERROR(EINVAL);  
        }  
        //Key Frame?  
        flags |= pkt->flags & AV_PKT_FLAG_KEY ? FLV_FRAME_KEY : FLV_FRAME_INTER;  
        break;  
    case AVMEDIA_TYPE_AUDIO:  

        flags = get_audio_flags(s, enc);  

        av_assert0(size);  
        //Type  
        avio_w8(pb, FLV_TAG_TYPE_AUDIO);  
        break;  
    case AVMEDIA_TYPE_DATA:  
        //Type  
        avio_w8(pb, FLV_TAG_TYPE_META);  
        break;  
    default:  
        return AVERROR(EINVAL);  
    }  

    if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4) {  
        /* check if extradata looks like mp4 formated */  
        if (enc->extradata_size > 0 && *(uint8_t*)enc->extradata != 1)  
            if ((ret = ff_avc_parse_nal_units_buf(pkt->data, &data, &size)) < 0)  
                return ret;  
    } else if (enc->codec_id == AV_CODEC_ID_AAC && pkt->size > 2 &&  
               (AV_RB16(pkt->data) & 0xfff0) == 0xfff0) {  
        if (!s->streams[pkt->stream_index]->nb_frames) {  
        av_log(s, AV_LOG_ERROR, "Malformed AAC bitstream detected: "  
               "use the audio bitstream filter 'aac_adtstoasc' to fix it "  
               "('-bsf:a aac_adtstoasc' option with ffmpeg)\n");  
        return AVERROR_INVALIDDATA;  
        }  
        av_log(s, AV_LOG_WARNING, "aac bitstream error\n");  
    }  

    /* check Speex packet duration */  
    if (enc->codec_id == AV_CODEC_ID_SPEEX && ts - sc->last_ts > 160)  
        av_log(s, AV_LOG_WARNING, "Warning: Speex stream has more than "  
                                  "8 frames per packet. Adobe Flash "  
                                  "Player cannot handle this!\n");  

    if (sc->last_ts < ts)  
        sc->last_ts = ts;  

    if (size + flags_size >= 1<<24) {  
        av_log(s, AV_LOG_ERROR, "Too large packet with size %u >= %u\n",  
               size + flags_size, 1<<24);  
        return AVERROR(EINVAL);  
    }  
    //Tag Header - Datasize  
    avio_wb24(pb, size + flags_size);  
    //Tag Header - Timestamp  
    avio_wb24(pb, ts & 0xFFFFFF);  
    avio_w8(pb, (ts >> 24) & 0x7F); // timestamps are 32 bits _signed_  
    //StreamID  
    avio_wb24(pb, flv->reserved);  

    if (enc->codec_type == AVMEDIA_TYPE_DATA) {  
        int data_size;  
        int64_t metadata_size_pos = avio_tell(pb);  
        if (enc->codec_id == AV_CODEC_ID_TEXT) {  
            // legacy FFmpeg magic?  
            avio_w8(pb, AMF_DATA_TYPE_STRING);  
            put_amf_string(pb, "onTextData");  
            avio_w8(pb, AMF_DATA_TYPE_MIXEDARRAY);  
            avio_wb32(pb, 2);  
            put_amf_string(pb, "type");  
            avio_w8(pb, AMF_DATA_TYPE_STRING);  
            put_amf_string(pb, "Text");  
            put_amf_string(pb, "text");  
            avio_w8(pb, AMF_DATA_TYPE_STRING);  
            put_amf_string(pb, pkt->data);  
            put_amf_string(pb, "");  
            avio_w8(pb, AMF_END_OF_OBJECT);  
        } else {  
            // just pass the metadata through  
            avio_write(pb, data ? data : pkt->data, size);  
        }  
        /* write total size of tag */  
        data_size = avio_tell(pb) - metadata_size_pos;  
        avio_seek(pb, metadata_size_pos - 10, SEEK_SET);  
        avio_wb24(pb, data_size);  
        avio_seek(pb, data_size + 10 - 3, SEEK_CUR);  
        avio_wb32(pb, data_size + 11);  
    } else {  
        av_assert1(flags>=0);  
        //First Byte of Tag Data  
        avio_w8(pb,flags);  
        if (enc->codec_id == AV_CODEC_ID_VP6)  
            avio_w8(pb,0);  
        if (enc->codec_id == AV_CODEC_ID_VP6F || enc->codec_id == AV_CODEC_ID_VP6A) {  
            if (enc->extradata_size)  
                avio_w8(pb, enc->extradata[0]);  
            else  
                avio_w8(pb, ((FFALIGN(enc->width,  16) - enc->width) << 4) |  
                             (FFALIGN(enc->height, 16) - enc->height));  
        } else if (enc->codec_id == AV_CODEC_ID_AAC)  
            avio_w8(pb, 1); // AAC raw  
        else if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4) {  
            //AVCVIDEOPACKET-AVCPacketType  
            avio_w8(pb, 1); // AVC NALU  
            //AVCVIDEOPACKET-CompositionTime  
            avio_wb24(pb, pkt->pts - pkt->dts);  
        }  
        //Data  
        avio_write(pb, data ? data : pkt->data, size);  

        avio_wb32(pb, size + flags_size + 11); // previous tag size  
        flv->duration = FFMAX(flv->duration,  
                              pkt->pts + flv->delay + pkt->duration);  
    }  

    av_free(data);  

    return pb->error;  
}

我们通过源代码简单梳理一下flv_write_packet()在写入H.264/AAC时候的流程： （1）写入Tag Header的Type，如果是视频，代码如下：

avio_w8(pb, FLV_TAG_TYPE_VIDEO);

如果是音频，代码如下：

avio_w8(pb, FLV_TAG_TYPE_AUDIO);

（2）写入Tag Header的Datasize，Timestamp和StreamID（至此完成Tag Header）：

//Tag Header - Datasize  
avio_wb24(pb, size + flags_size);  
//Tag Header - Timestamp  
avio_wb24(pb, ts & 0xFFFFFF);  
avio_w8(pb, (ts >> 24) & 0x7F); // timestamps are 32 bits _signed_  
//StreamID  
avio_wb24(pb, flv->reserved);

（3）写入Tag Data的第一字节（其中flag已经在前面的代码中设置完毕）：

//First Byte of Tag Data  
avio_w8(pb,flags);

（4）如果编码格式VP6作相应的处理（不研究）；编码格式为AAC，写入AACAUDIODATA；编码格式为H.264，写入AVCVIDEOPACKET：

if (enc->codec_id == AV_CODEC_ID_VP6F || enc->codec_id == AV_CODEC_ID_VP6A) {  
    if (enc->extradata_size)  
        avio_w8(pb, enc->extradata[0]);  
    else  
        avio_w8(pb, ((FFALIGN(enc->width,  16) - enc->width) << 4) |  
                     (FFALIGN(enc->height, 16) - enc->height));  
} else if (enc->codec_id == AV_CODEC_ID_AAC)  
    avio_w8(pb, 1); // AAC raw  
else if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4) {  
    //AVCVIDEOPACKET-AVCPacketType  
    avio_w8(pb, 1); // AVC NALU  
    //AVCVIDEOPACKET-CompositionTime  
    avio_wb24(pb, pkt->pts - pkt->dts);  
}

（5）写入数据：

//Data  
avio_write(pb, data ? data : pkt->data, size);

（6） 写入previous tag size：

avio_wb32(pb, size + flags_size + 11); // previous tag size

至此，flv_write_packet()就完成了一个Tag的写入