FFMPEG中关于ts流的时长估计的实现(转)

最近在做H.265 编码，原本只是做编码器的实现，但客户项目涉及到ts的封装，搞得我不得不配合了解点ts方面的东西。下面技术文档不错，转一下。
ts流中的时间估计

我们知道ts流中是没有时间信息的，我门来看看ffmpeg是怎么估计其duration的

方法1.通过pts来估计

static void estimate_timings_from_pts(AVFormatContext *ic, int64_t old_offset)

{

    AVPacket pkt1, *pkt = &pkt1;

    AVStream *st;

    int read_size, i, ret;

    int64_t end_time;

    int64_t filesize, offset, duration;

    int retry=0;

    /* flush packet queue */

    flush_packet_queue(ic);

    for (i=0; i<ic->nb_streams; i++) {

        st = ic->streams[i];

        if (st->start_time == AV_NOPTS_VALUE && st->first_dts == AV_NOPTS_VALUE)

            av_log(st->codec, AV_LOG_WARNING, "start time is not set in estimate_timings_from_pts\n");

        if (st->parser) {

            av_parser_close(st->parser);

            st->parser= NULL;

        }

    }

    /* estimate the end time (duration) */

    /* XXX: may need to support wrapping */

    filesize = ic->pb ? avio_size(ic->pb) : 0;//得到文件大小

    end_time = AV_NOPTS_VALUE;

    do{

        offset = filesize - (DURATION_MAX_READ_SIZE<<retry);

        if (offset < 0)

            offset = 0;

        avio_seek(ic->pb, offset, SEEK_SET);//尽量往后查找，pts越靠近文件末尾，利用pts估计时长越准确

        read_size = 0;

        for(;;) {

            if (read_size >= DURATION_MAX_READ_SIZE<<(FFMAX(retry-1,0)))

                break;

            do {

                ret = ff_read_packet(ic, pkt);//从接近文件末尾的地方读取数据，直到最后一个合法的数据包

            } while(ret == AVERROR(EAGAIN));

            if (ret != 0)

                break;

            read_size += pkt->size;

            st = ic->streams[pkt->stream_index];

            if (pkt->pts != AV_NOPTS_VALUE &&

                (st->start_time != AV_NOPTS_VALUE ||

                 st->first_dts  != AV_NOPTS_VALUE)) {

                duration = end_time = pkt->pts;//利用该包的pts数据得到比较接近的时长

                if (st->start_time != AV_NOPTS_VALUE)

                    duration -= st->start_time;//减去初始时间

                else

                    duration -= st->first_dts;

                if (duration > 0) {

                    if (st->duration == AV_NOPTS_VALUE || st->duration < duration)

                        st->duration = duration;

                }

            }

            av_free_packet(pkt);

        }

    }while(   end_time==AV_NOPTS_VALUE

           && filesize > (DURATION_MAX_READ_SIZE<<retry)

           && ++retry <= DURATION_MAX_RETRY);//尝试 DURATION_MAX_RETRY这么多次

}

方法2：通过文件大小和码流来估计

static void estimate_timings_from_bit_rate(AVFormatContext *ic)

{

    int64_t filesize, duration;

    int bit_rate, i;

    AVStream *st;

    /* if bit_rate is already set, we believe it */

    if (ic->bit_rate <= 0) {

        bit_rate = 0;

        for(i=0;i<ic->nb_streams;i++) {//通过累积各个子流的平均码率得到文件的平均码率

            st = ic->streams[i];

            if (st->codec->bit_rate > 0)

            bit_rate += st->codec->bit_rate;

        }

        ic->bit_rate = bit_rate;

    }

    /* if duration is already set, we believe it */

    if (ic->duration == AV_NOPTS_VALUE &&

        ic->bit_rate != 0) {

        filesize = ic->pb ? avio_size(ic->pb) : 0;

        if (filesize > 0) {

            for(i = 0; i < ic->nb_streams; i++) {

                st = ic->streams[i];

                duration= av_rescale(8*filesize, st->time_base.den, ic->bit_rate*(int64_t)st->time_base.num);//通过文件大小除以文件平均码率得到文件时长，之所以还有time_base信息，是因为最后要把秒转换为以time_base为单位的值

                if (st->duration == AV_NOPTS_VALUE)

                    st->duration = duration;

            }

        }

    }

}

对应的，也可以在android stagefright中加入类似的实现：

uint64_t MPEG2TSExtractor::estimateDuration() {

    Mutex::Autolock autoLock(mLock);

    int64_t filesize;

    int64_t end_time;

    int64_t  offset, duration;

    int retry=1;

    status_t re=mDataSource->getSize(&filesize);//android中有类似的函数去得到文件的大小

    if (re != OK) {

        ALOGE("Failed to get file size");

        return ERROR_MALFORMED;

    }

    uint8_t packet[kTSPacketSize];

    unsigned payload_unit_start_indicator = 0;

    unsigned PID = 0;

    unsigned adaptation_field_control = 0;

//实现思想：从文件末尾开始读取188个字节，直到找到第一个pes包的边界，并且跳过adp filed的包，这里认为adp field不含有合法的pts

    while (1) {

        offset = filesize - kTSPacketSize*retry;

        ssize_t n = mDataSource->readAt(offset, packet, kTSPacketSize);

        if (n < (ssize_t)kTSPacketSize) {

            return (n < 0) ? (status_t)n : ERROR_END_OF_STREAM;

        }

        ABitReader* br= new ABitReader((const uint8_t *)packet, kTSPacketSize);//主要此类实现了一个bit读取器，对于码流的解析非常方便，类似于ffmpeg中的get_bits.h中实现的功能

        unsigned sync_byte = br->getBits(8);

        CHECK_EQ(sync_byte, 0x47u);

        br->skipBits(1);

        unsigned payload_unit_start_indicator = br->getBits(1);

        br->skipBits(1);

        PID = br->getBits(13);

        br->skipBits(2);

        adaptation_field_control = br->getBits(2);

        if ((payload_unit_start_indicator == 1) && (adaptation_field_control == 1) &&

            (PID != 0x00u) && (PID != 0x01u) && (PID != 0x02u) ) {

            break;

        }

        retry++;

        delete br;

    } ;

    ABitReader* br= new ABitReader((const uint8_t *)packet, kTSPacketSize);

    br->skipBits(8 + 3 + 13);

    br->skipBits(2);

    adaptation_field_control = br->getBits(2);

    ALOGV("adaptation_field_control = %u", adaptation_field_control);

    br->skipBits(4);

    if (adaptation_field_control == 2 || adaptation_field_control == 3) {

       unsigned adaptation_field_length = br->getBits(8);

       if (adaptation_field_length > 0) {

           br->skipBits(adaptation_field_length * 8);

       }

       ALOGV("adaptation_field_length = %u", adaptation_field_length);

    }

    if (adaptation_field_control == 1 || adaptation_field_control == 3) {

        unsigned packet_startcode_prefix = br->getBits(24);

        ALOGV("packet_startcode_prefix = 0x%08x", packet_startcode_prefix);

        CHECK_EQ(packet_startcode_prefix, 0x000001u);

        unsigned stream_id = br->getBits(8);

        ALOGV("stream_id = 0x%02x", stream_id);

        br->skipBits(16);

        //以下可以参考标准，标准上解释的很详细，应该不难理解

        if (stream_id != 0xbc  // program_stream_map

                && stream_id != 0xbe  // padding_stream

                && stream_id != 0xbf  // private_stream_2

                && stream_id != 0xf0  // ECM

                && stream_id != 0xf1  // EMM

                && stream_id != 0xff  // program_stream_directory

                && stream_id != 0xf2  // DSMCC

                && stream_id != 0xf8) {  // H.222.1 type E

              CHECK_EQ(br->getBits(2), 2u);

            br->skipBits(6);

            unsigned PTS_DTS_flags = br->getBits(2);

            ALOGV("PTS_DTS_flags = %u", PTS_DTS_flags);

            br->skipBits(6);

            unsigned PES_header_data_length = br->getBits(8);

            unsigned optional_bytes_remaining = PES_header_data_length;

            uint64_t PTS = 0, DTS = 0;

            if (PTS_DTS_flags == 2 || PTS_DTS_flags == 3) {

                CHECK_GE(optional_bytes_remaining, 5u);

                CHECK_EQ(br->getBits(4), PTS_DTS_flags);

                PTS = ((uint64_t)br->getBits(3)) << 30;

                CHECK_EQ(br->getBits(1), 1u);

                PTS |= ((uint64_t)br->getBits(15)) << 15;

                CHECK_EQ(br->getBits(1), 1u);

                PTS |= br->getBits(15);

                CHECK_EQ(br->getBits(1), 1u);

                ALOGV("PTS = %llu", PTS);

                ALOGV("PTS = %.2f secs", PTS / 90000.0f);

                PTS = (PTS * 1000 * 1000ll) / 90000;

                return PTS;

            }

             // ES data follows.

        }

    }

    delete br;

    return 0;

}

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FFMPEG中关于ts流的时长估计的实现(转)

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