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C++ put_bits函数代码示例

原作者: [db:作者] 来自: [db:来源] 收藏 邀请

本文整理汇总了C++中put_bits函数的典型用法代码示例。如果您正苦于以下问题:C++ put_bits函数的具体用法?C++ put_bits怎么用?C++ put_bits使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。



在下文中一共展示了put_bits函数的20个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于我们的系统推荐出更棒的C++代码示例。

示例1: put_sample7

static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
                               int scale_factor)
{
    sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
    put_bits(&c->pb, bits, quantize((int) sample, bits));
}
开发者ID:mcodegeeks,项目名称:OpenKODE-Framework,代码行数:6,代码来源:dcaenc.cpp


示例2: ff_h263_encode_picture_header

void ff_h263_encode_picture_header(MpegEncContext * s, int picture_number)
{
    int format, coded_frame_rate, coded_frame_rate_base, i, temp_ref;
    int best_clock_code=1;
    int best_divisor=60;
    int best_error= INT_MAX;

    if(s->h263_plus){
        for(i=0; i<2; i++){
            int div, error;
            div= (s->avctx->time_base.num*1800000LL + 500LL*s->avctx->time_base.den) / ((1000LL+i)*s->avctx->time_base.den);
            div= av_clip(div, 1, 127);
            error= FFABS(s->avctx->time_base.num*1800000LL - (1000LL+i)*s->avctx->time_base.den*div);
            if(error < best_error){
                best_error= error;
                best_divisor= div;
                best_clock_code= i;
            }
        }
    }
    s->custom_pcf= best_clock_code!=1 || best_divisor!=60;
    coded_frame_rate= 1800000;
    coded_frame_rate_base= (1000+best_clock_code)*best_divisor;

    avpriv_align_put_bits(&s->pb);

    /* Update the pointer to last GOB */
    s->ptr_lastgob = put_bits_ptr(&s->pb);
    put_bits(&s->pb, 22, 0x20); /* PSC */
    temp_ref= s->picture_number * (int64_t)coded_frame_rate * s->avctx->time_base.num / //FIXME use timestamp
                         (coded_frame_rate_base * (int64_t)s->avctx->time_base.den);
    put_sbits(&s->pb, 8, temp_ref); /* TemporalReference */

    put_bits(&s->pb, 1, 1);     /* marker */
    put_bits(&s->pb, 1, 0);     /* h263 id */
    put_bits(&s->pb, 1, 0);     /* split screen off */
    put_bits(&s->pb, 1, 0);     /* camera  off */
    put_bits(&s->pb, 1, 0);     /* freeze picture release off */

    format = ff_match_2uint16(ff_h263_format, FF_ARRAY_ELEMS(ff_h263_format), s->width, s->height);
    if (!s->h263_plus) {
        /* H.263v1 */
        put_bits(&s->pb, 3, format);
        put_bits(&s->pb, 1, (s->pict_type == AV_PICTURE_TYPE_P));
        /* By now UMV IS DISABLED ON H.263v1, since the restrictions
        of H.263v1 UMV implies to check the predicted MV after
        calculation of the current MB to see if we're on the limits */
        put_bits(&s->pb, 1, 0);         /* Unrestricted Motion Vector: off */
        put_bits(&s->pb, 1, 0);         /* SAC: off */
        put_bits(&s->pb, 1, s->obmc);   /* Advanced Prediction */
        put_bits(&s->pb, 1, 0);         /* only I/P frames, no PB frame */
        put_bits(&s->pb, 5, s->qscale);
        put_bits(&s->pb, 1, 0);         /* Continuous Presence Multipoint mode: off */
    } else {
        int ufep=1;
        /* H.263v2 */
        /* H.263 Plus PTYPE */

        put_bits(&s->pb, 3, 7);
        put_bits(&s->pb,3,ufep); /* Update Full Extended PTYPE */
        if (format == 8)
            put_bits(&s->pb,3,6); /* Custom Source Format */
        else
            put_bits(&s->pb, 3, format);

        put_bits(&s->pb,1, s->custom_pcf);
        put_bits(&s->pb,1, s->umvplus); /* Unrestricted Motion Vector */
        put_bits(&s->pb,1,0); /* SAC: off */
        put_bits(&s->pb,1,s->obmc); /* Advanced Prediction Mode */
        put_bits(&s->pb,1,s->h263_aic); /* Advanced Intra Coding */
        put_bits(&s->pb,1,s->loop_filter); /* Deblocking Filter */
        put_bits(&s->pb,1,s->h263_slice_structured); /* Slice Structured */
        put_bits(&s->pb,1,0); /* Reference Picture Selection: off */
        put_bits(&s->pb,1,0); /* Independent Segment Decoding: off */
        put_bits(&s->pb,1,s->alt_inter_vlc); /* Alternative Inter VLC */
        put_bits(&s->pb,1,s->modified_quant); /* Modified Quantization: */
        put_bits(&s->pb,1,1); /* "1" to prevent start code emulation */
        put_bits(&s->pb,3,0); /* Reserved */

        put_bits(&s->pb, 3, s->pict_type == AV_PICTURE_TYPE_P);

        put_bits(&s->pb,1,0); /* Reference Picture Resampling: off */
        put_bits(&s->pb,1,0); /* Reduced-Resolution Update: off */
        put_bits(&s->pb,1,s->no_rounding); /* Rounding Type */
        put_bits(&s->pb,2,0); /* Reserved */
        put_bits(&s->pb,1,1); /* "1" to prevent start code emulation */

        /* This should be here if PLUSPTYPE */
        put_bits(&s->pb, 1, 0); /* Continuous Presence Multipoint mode: off */

        if (format == 8) {
            /* Custom Picture Format (CPFMT) */
            s->aspect_ratio_info= ff_h263_aspect_to_info(s->avctx->sample_aspect_ratio);

            put_bits(&s->pb,4,s->aspect_ratio_info);
            put_bits(&s->pb,9,(s->width >> 2) - 1);
            put_bits(&s->pb,1,1); /* "1" to prevent start code emulation */
            put_bits(&s->pb,9,(s->height >> 2));
            if (s->aspect_ratio_info == FF_ASPECT_EXTENDED){
                put_bits(&s->pb, 8, s->avctx->sample_aspect_ratio.num);
//.........这里部分代码省略.........
开发者ID:57-Wolve,项目名称:FFmpeg,代码行数:101,代码来源:ituh263enc.c


示例3: ff_wmv2_encode_picture_header

int ff_wmv2_encode_picture_header(MpegEncContext * s, int picture_number)
{
    Wmv2Context * const w= (Wmv2Context*)s;

    put_bits(&s->pb, 1, s->pict_type - 1);
    if(s->pict_type == FF_I_TYPE){
        put_bits(&s->pb, 7, 0);
    }
    put_bits(&s->pb, 5, s->qscale);

    s->dc_table_index = 1;
    s->mv_table_index = 1; /* only if P frame */
    s->per_mb_rl_table = 0;
    s->mspel= 0;
    w->per_mb_abt=0;
    w->abt_type=0;
    w->j_type=0;

    assert(s->flipflop_rounding);

    if (s->pict_type == FF_I_TYPE) {
        assert(s->no_rounding==1);
        if(w->j_type_bit) put_bits(&s->pb, 1, w->j_type);

        if(w->per_mb_rl_bit) put_bits(&s->pb, 1, s->per_mb_rl_table);

        if(!s->per_mb_rl_table){
            ff_msmpeg4_code012(&s->pb, s->rl_chroma_table_index);
            ff_msmpeg4_code012(&s->pb, s->rl_table_index);
        }

        put_bits(&s->pb, 1, s->dc_table_index);

        s->inter_intra_pred= 0;
    }else{
        int cbp_index;

        put_bits(&s->pb, 2, SKIP_TYPE_NONE);

        ff_msmpeg4_code012(&s->pb, cbp_index=0);
        if(s->qscale <= 10){
            int map[3]= {0,2,1};
            w->cbp_table_index= map[cbp_index];
        }else if(s->qscale <= 20){
            int map[3]= {1,0,2};
            w->cbp_table_index= map[cbp_index];
        }else{
            int map[3]= {2,1,0};
            w->cbp_table_index= map[cbp_index];
        }

        if(w->mspel_bit) put_bits(&s->pb, 1, s->mspel);

        if(w->abt_flag){
            put_bits(&s->pb, 1, w->per_mb_abt^1);
            if(!w->per_mb_abt){
                ff_msmpeg4_code012(&s->pb, w->abt_type);
            }
        }

        if(w->per_mb_rl_bit) put_bits(&s->pb, 1, s->per_mb_rl_table);

        if(!s->per_mb_rl_table){
            ff_msmpeg4_code012(&s->pb, s->rl_table_index);
            s->rl_chroma_table_index = s->rl_table_index;
        }
        put_bits(&s->pb, 1, s->dc_table_index);
        put_bits(&s->pb, 1, s->mv_table_index);

        s->inter_intra_pred= 0;//(s->width*s->height < 320*240 && s->bit_rate<=II_BITRATE);
    }
    s->esc3_level_length= 0;
    s->esc3_run_length= 0;

    return 0;
}
开发者ID:JDsolution,项目名称:ipnc,代码行数:76,代码来源:wmv2enc.c


示例4: flv_write_codec_header

static void flv_write_codec_header(AVFormatContext* s, AVCodecParameters* par) {
    int64_t data_size;
    AVIOContext *pb = s->pb;
    FLVContext *flv = s->priv_data;

    if (par->codec_id == AV_CODEC_ID_AAC || par->codec_id == AV_CODEC_ID_H264
            || par->codec_id == AV_CODEC_ID_MPEG4) {
        int64_t pos;
        avio_w8(pb,
                par->codec_type == AVMEDIA_TYPE_VIDEO ?
                        FLV_TAG_TYPE_VIDEO : FLV_TAG_TYPE_AUDIO);
        avio_wb24(pb, 0); // size patched later
        avio_wb24(pb, 0); // ts
        avio_w8(pb, 0);   // ts ext
        avio_wb24(pb, 0); // streamid
        pos = avio_tell(pb);
        if (par->codec_id == AV_CODEC_ID_AAC) {
            avio_w8(pb, get_audio_flags(s, par));
            avio_w8(pb, 0); // AAC sequence header

            if (!par->extradata_size && (flv->flags & FLV_AAC_SEQ_HEADER_DETECT)) {
                PutBitContext pbc;
                int samplerate_index;
                int channels = flv->audio_par->channels
                        - (flv->audio_par->channels == 8 ? 1 : 0);
                uint8_t data[2];

                for (samplerate_index = 0; samplerate_index < 16;
                        samplerate_index++)
                    if (flv->audio_par->sample_rate
                            == mpeg4audio_sample_rates[samplerate_index])
                        break;

                init_put_bits(&pbc, data, sizeof(data));
                put_bits(&pbc, 5, flv->audio_par->profile + 1); //profile
                put_bits(&pbc, 4, samplerate_index); //sample rate index
                put_bits(&pbc, 4, channels);
                put_bits(&pbc, 1, 0); //frame length - 1024 samples
                put_bits(&pbc, 1, 0); //does not depend on core coder
                put_bits(&pbc, 1, 0); //is not extension
                flush_put_bits(&pbc);

                avio_w8(pb, data[0]);
                avio_w8(pb, data[1]);

                av_log(s, AV_LOG_WARNING, "AAC sequence header: %02x %02x.\n",
                        data[0], data[1]);
            }
            avio_write(pb, par->extradata, par->extradata_size);
        } else {
            avio_w8(pb, par->codec_tag | FLV_FRAME_KEY); // flags
            avio_w8(pb, 0); // AVC sequence header
            avio_wb24(pb, 0); // composition time
            ff_isom_write_avcc(pb, par->extradata, par->extradata_size);
        }
        data_size = avio_tell(pb) - pos;
        avio_seek(pb, -data_size - 10, SEEK_CUR);
        avio_wb24(pb, data_size);
        avio_skip(pb, data_size + 10 - 3);
        avio_wb32(pb, data_size + 11); // previous tag size
    }
}
开发者ID:garybuhrmaster,项目名称:mythtv,代码行数:62,代码来源:flvenc.c


示例5: encode_block

static int encode_block(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
                        int total_gain)
{
    int v, bsize, ch, coef_nb_bits, parse_exponents;
    float mdct_norm;
    int nb_coefs[MAX_CHANNELS];
    static const int fixed_exp[25] = {
        20, 20, 20, 20, 20,
        20, 20, 20, 20, 20,
        20, 20, 20, 20, 20,
        20, 20, 20, 20, 20,
        20, 20, 20, 20, 20
    };

    // FIXME remove duplication relative to decoder
    if (s->use_variable_block_len) {
        assert(0); // FIXME not implemented
    } else {
        /* fixed block len */
        s->next_block_len_bits = s->frame_len_bits;
        s->prev_block_len_bits = s->frame_len_bits;
        s->block_len_bits      = s->frame_len_bits;
    }

    s->block_len = 1 << s->block_len_bits;
//     assert((s->block_pos + s->block_len) <= s->frame_len);
    bsize = s->frame_len_bits - s->block_len_bits;

    // FIXME factor
    v = s->coefs_end[bsize] - s->coefs_start;
    for (ch = 0; ch < s->avctx->channels; ch++)
        nb_coefs[ch] = v;
    {
        int n4 = s->block_len / 2;
        mdct_norm = 1.0 / (float) n4;
        if (s->version == 1)
            mdct_norm *= sqrt(n4);
    }

    if (s->avctx->channels == 2)
        put_bits(&s->pb, 1, !!s->ms_stereo);

    for (ch = 0; ch < s->avctx->channels; ch++) {
        // FIXME only set channel_coded when needed, instead of always
        s->channel_coded[ch] = 1;
        if (s->channel_coded[ch])
            init_exp(s, ch, fixed_exp);
    }

    for (ch = 0; ch < s->avctx->channels; ch++) {
        if (s->channel_coded[ch]) {
            WMACoef *coefs1;
            float *coefs, *exponents, mult;
            int i, n;

            coefs1    = s->coefs1[ch];
            exponents = s->exponents[ch];
            mult      = pow(10, total_gain * 0.05) / s->max_exponent[ch];
            mult     *= mdct_norm;
            coefs     = src_coefs[ch];
            if (s->use_noise_coding && 0) {
                assert(0); // FIXME not implemented
            } else {
                coefs += s->coefs_start;
                n      = nb_coefs[ch];
                for (i = 0; i < n; i++) {
                    double t = *coefs++ / (exponents[i] * mult);
                    if (t < -32768 || t > 32767)
                        return -1;

                    coefs1[i] = lrint(t);
                }
            }
        }
    }

    v = 0;
    for (ch = 0; ch < s->avctx->channels; ch++) {
        int a = s->channel_coded[ch];
        put_bits(&s->pb, 1, a);
        v |= a;
    }

    if (!v)
        return 1;

    for (v = total_gain - 1; v >= 127; v -= 127)
        put_bits(&s->pb, 7, 127);
    put_bits(&s->pb, 7, v);

    coef_nb_bits = ff_wma_total_gain_to_bits(total_gain);

    if (s->use_noise_coding) {
        for (ch = 0; ch < s->avctx->channels; ch++) {
            if (s->channel_coded[ch]) {
                int i, n;
                n = s->exponent_high_sizes[bsize];
                for (i = 0; i < n; i++) {
                    put_bits(&s->pb, 1, s->high_band_coded[ch][i] = 0);
                    if (0)
//.........这里部分代码省略.........
开发者ID:AVLeo,项目名称:libav,代码行数:101,代码来源:wmaenc.c


示例6: codebook_trellis_rate


//.........这里部分代码省略.........
        size = sce->ics.swb_sizes[swb];
        if (sce->zeroes[win*16 + swb]) {
            float cost_stay_here = path[swb][0].cost;
            float cost_get_here  = next_minbits + run_bits + 4;
            if (   run_value_bits[sce->ics.num_windows == 8][path[swb][0].run]
                != run_value_bits[sce->ics.num_windows == 8][path[swb][0].run+1])
                cost_stay_here += run_bits;
            if (cost_get_here < cost_stay_here) {
                path[swb+1][0].prev_idx = next_mincb;
                path[swb+1][0].cost     = cost_get_here;
                path[swb+1][0].run      = 1;
            } else {
                path[swb+1][0].prev_idx = 0;
                path[swb+1][0].cost     = cost_stay_here;
                path[swb+1][0].run      = path[swb][0].run + 1;
            }
            next_minbits = path[swb+1][0].cost;
            next_mincb = 0;
            for (cb = 1; cb < 12; cb++) {
                path[swb+1][cb].cost = 61450;
                path[swb+1][cb].prev_idx = -1;
                path[swb+1][cb].run = 0;
            }
        } else {
            float minbits = next_minbits;
            int mincb = next_mincb;
            int startcb = sce->band_type[win*16+swb];
            next_minbits = INFINITY;
            next_mincb = 0;
            for (cb = 0; cb < startcb; cb++) {
                path[swb+1][cb].cost = 61450;
                path[swb+1][cb].prev_idx = -1;
                path[swb+1][cb].run = 0;
            }
            for (cb = startcb; cb < 12; cb++) {
                float cost_stay_here, cost_get_here;
                float bits = 0.0f;
                for (w = 0; w < group_len; w++) {
                    bits += quantize_band_cost(s, sce->coeffs + start + w*128,
                                               s->scoefs + start + w*128, size,
                                               sce->sf_idx[(win+w)*16+swb], cb,
                                               0, INFINITY, NULL);
                }
                cost_stay_here = path[swb][cb].cost + bits;
                cost_get_here  = minbits            + bits + run_bits + 4;
                if (   run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
                    != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
                    cost_stay_here += run_bits;
                if (cost_get_here < cost_stay_here) {
                    path[swb+1][cb].prev_idx = mincb;
                    path[swb+1][cb].cost     = cost_get_here;
                    path[swb+1][cb].run      = 1;
                } else {
                    path[swb+1][cb].prev_idx = cb;
                    path[swb+1][cb].cost     = cost_stay_here;
                    path[swb+1][cb].run      = path[swb][cb].run + 1;
                }
                if (path[swb+1][cb].cost < next_minbits) {
                    next_minbits = path[swb+1][cb].cost;
                    next_mincb = cb;
                }
            }
        }
        start += sce->ics.swb_sizes[swb];
    }

    //convert resulting path from backward-linked list
    stack_len = 0;
    idx       = 0;
    for (cb = 1; cb < 12; cb++)
        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
            idx = cb;
    ppos = max_sfb;
    while (ppos > 0) {
        assert(idx >= 0);
        cb = idx;
        stackrun[stack_len] = path[ppos][cb].run;
        stackcb [stack_len] = cb;
        idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
        ppos -= path[ppos][cb].run;
        stack_len++;
    }
    //perform actual band info encoding
    start = 0;
    for (i = stack_len - 1; i >= 0; i--) {
        put_bits(&s->pb, 4, stackcb[i]);
        count = stackrun[i];
        memset(sce->zeroes + win*16 + start, !stackcb[i], count);
        //XXX: memset when band_type is also uint8_t
        for (j = 0; j < count; j++) {
            sce->band_type[win*16 + start] =  stackcb[i];
            start++;
        }
        while (count >= run_esc) {
            put_bits(&s->pb, run_bits, run_esc);
            count -= run_esc;
        }
        put_bits(&s->pb, run_bits, count);
    }
}
开发者ID:AVbin,项目名称:libav,代码行数:101,代码来源:aaccoder.c


示例7: ff_flv_encode_picture_header

void ff_flv_encode_picture_header(MpegEncContext * s, int picture_number)
{
      int format;

      avpriv_align_put_bits(&s->pb);

      put_bits(&s->pb, 17, 1);
      put_bits(&s->pb, 5, (s->h263_flv-1)); /* 0: h263 escape codes 1: 11-bit escape codes */
      put_bits(&s->pb, 8, (((int64_t)s->picture_number * 30 * s->avctx->time_base.num) / //FIXME use timestamp
                           s->avctx->time_base.den) & 0xff); /* TemporalReference */
      if (s->width == 352 && s->height == 288)
        format = 2;
      else if (s->width == 176 && s->height == 144)
        format = 3;
      else if (s->width == 128 && s->height == 96)
        format = 4;
      else if (s->width == 320 && s->height == 240)
        format = 5;
      else if (s->width == 160 && s->height == 120)
        format = 6;
      else if (s->width <= 255 && s->height <= 255)
        format = 0; /* use 1 byte width & height */
      else
        format = 1; /* use 2 bytes width & height */
      put_bits(&s->pb, 3, format); /* PictureSize */
      if (format == 0) {
        put_bits(&s->pb, 8, s->width);
        put_bits(&s->pb, 8, s->height);
      } else if (format == 1) {
        put_bits(&s->pb, 16, s->width);
        put_bits(&s->pb, 16, s->height);
      }
      put_bits(&s->pb, 2, s->pict_type == AV_PICTURE_TYPE_P); /* PictureType */
      put_bits(&s->pb, 1, 1); /* DeblockingFlag: on */
      put_bits(&s->pb, 5, s->qscale); /* Quantizer */
      put_bits(&s->pb, 1, 0); /* ExtraInformation */

      if(s->h263_aic){
        s->y_dc_scale_table=
          s->c_dc_scale_table= ff_aic_dc_scale_table;
      }else{
        s->y_dc_scale_table=
          s->c_dc_scale_table= ff_mpeg1_dc_scale_table;
      }
}
开发者ID:Alcantor,项目名称:FFmpeg,代码行数:45,代码来源:flvenc.c


示例8: swf_write_header

static int swf_write_header(AVFormatContext *s)
{
    SWFContext *swf = s->priv_data;
    AVIOContext *pb = s->pb;
    PutBitContext p;
    uint8_t buf1[256];
    int i, width, height, rate, rate_base;
    int version;

    swf->sound_samples = 0;
    swf->swf_frame_number = 0;
    swf->video_frame_number = 0;

    for(i=0;i<s->nb_streams;i++) {
        AVCodecContext *enc = s->streams[i]->codec;
        if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
            if (enc->codec_id == AV_CODEC_ID_MP3) {
                swf->audio_enc = enc;
                swf->audio_fifo= av_fifo_alloc(AUDIO_FIFO_SIZE);
                if (!swf->audio_fifo)
                    return AVERROR(ENOMEM);
            } else {
                av_log(s, AV_LOG_ERROR, "SWF muxer only supports MP3\n");
                return -1;
            }
        } else {
            if (enc->codec_id == AV_CODEC_ID_VP6F ||
                enc->codec_id == AV_CODEC_ID_FLV1 ||
                enc->codec_id == AV_CODEC_ID_MJPEG) {
                swf->video_enc = enc;
            } else {
                av_log(s, AV_LOG_ERROR, "SWF muxer only supports VP6, FLV1 and MJPEG\n");
                return -1;
            }
        }
    }

    if (!swf->video_enc) {
        /* currently, cannot work correctly if audio only */
        width = 320;
        height = 200;
        rate = 10;
        rate_base= 1;
    } else {
        width = swf->video_enc->width;
        height = swf->video_enc->height;
        rate = swf->video_enc->time_base.den;
        rate_base = swf->video_enc->time_base.num;
    }

    if (!swf->audio_enc)
        swf->samples_per_frame = (44100. * rate_base) / rate;
    else
        swf->samples_per_frame = (swf->audio_enc->sample_rate * rate_base) / rate;

    avio_write(pb, "FWS", 3);

    if (!strcmp("avm2", s->oformat->name))
        version = 9;
    else if (swf->video_enc && swf->video_enc->codec_id == AV_CODEC_ID_VP6F)
        version = 8; /* version 8 and above support VP6 codec */
    else if (swf->video_enc && swf->video_enc->codec_id == AV_CODEC_ID_FLV1)
        version = 6; /* version 6 and above support FLV1 codec */
    else
        version = 4; /* version 4 for mpeg audio support */
    avio_w8(pb, version);

    avio_wl32(pb, DUMMY_FILE_SIZE); /* dummy size
                                      (will be patched if not streamed) */

    put_swf_rect(pb, 0, width * 20, 0, height * 20);
    avio_wl16(pb, (rate * 256) / rate_base); /* frame rate */
    swf->duration_pos = avio_tell(pb);
    avio_wl16(pb, (uint16_t)(DUMMY_DURATION * (int64_t)rate / rate_base)); /* frame count */

    /* avm2/swf v9 (also v8?) files require a file attribute tag */
    if (version == 9) {
        put_swf_tag(s, TAG_FILEATTRIBUTES);
        avio_wl32(pb, 1<<3); /* set ActionScript v3/AVM2 flag */
        put_swf_end_tag(s);
    }

    /* define a shape with the jpeg inside */
    if (swf->video_enc && swf->video_enc->codec_id == AV_CODEC_ID_MJPEG) {
        put_swf_tag(s, TAG_DEFINESHAPE);

        avio_wl16(pb, SHAPE_ID); /* ID of shape */
        /* bounding rectangle */
        put_swf_rect(pb, 0, width, 0, height);
        /* style info */
        avio_w8(pb, 1); /* one fill style */
        avio_w8(pb, 0x41); /* clipped bitmap fill */
        avio_wl16(pb, BITMAP_ID); /* bitmap ID */
        /* position of the bitmap */
        put_swf_matrix(pb, (int)(1.0 * (1 << FRAC_BITS)), 0,
                       0, (int)(1.0 * (1 << FRAC_BITS)), 0, 0);
        avio_w8(pb, 0); /* no line style */

        /* shape drawing */
        init_put_bits(&p, buf1, sizeof(buf1));
//.........这里部分代码省略.........
开发者ID:Dahlqvist,项目名称:sfeMovie,代码行数:101,代码来源:swfenc.c


示例9: ff_adts_write_frame_header

int ff_adts_write_frame_header(ADTSContext *ctx,
                               uint8_t *buf, int size, int pce_size)
{
    PutBitContext pb;

    unsigned full_frame_size = (unsigned)ADTS_HEADER_SIZE + size + pce_size;
    if (full_frame_size > ADTS_MAX_FRAME_BYTES) {
        av_log(NULL, AV_LOG_ERROR, "ADTS frame size too large: %u (max %d)\n",
               full_frame_size, ADTS_MAX_FRAME_BYTES);
        return AVERROR_INVALIDDATA;
    }

    init_put_bits(&pb, buf, ADTS_HEADER_SIZE);

    /* adts_fixed_header */
    put_bits(&pb, 12, 0xfff);   /* syncword */
    put_bits(&pb, 1, 0);        /* ID */
    put_bits(&pb, 2, 0);        /* layer */
    put_bits(&pb, 1, 1);        /* protection_absent */
    put_bits(&pb, 2, ctx->objecttype); /* profile_objecttype */
    put_bits(&pb, 4, ctx->sample_rate_index);
    put_bits(&pb, 1, 0);        /* private_bit */
    put_bits(&pb, 3, ctx->channel_conf); /* channel_configuration */
    put_bits(&pb, 1, 0);        /* original_copy */
    put_bits(&pb, 1, 0);        /* home */

    /* adts_variable_header */
    put_bits(&pb, 1, 0);        /* copyright_identification_bit */
    put_bits(&pb, 1, 0);        /* copyright_identification_start */
    put_bits(&pb, 13, full_frame_size); /* aac_frame_length */
    put_bits(&pb, 11, 0x7ff);   /* adts_buffer_fullness */
    put_bits(&pb, 2, 0);        /* number_of_raw_data_blocks_in_frame */

    flush_put_bits(&pb);

    return 0;
}
开发者ID:stainberg,项目名称:android_FFMPEG,代码行数:37,代码来源:adtsenc.c


示例10: put_swf_line_edge

static void put_swf_line_edge(PutBitContext *pb, int dx, int dy)
{
    int nbits, mask;

    put_bits(pb, 1, 1); /* edge */
    put_bits(pb, 1, 1); /* line select */
    nbits = 2;
    max_nbits(&nbits, dx);
    max_nbits(&nbits, dy);

    mask = (1 << nbits) - 1;
    put_bits(pb, 4, nbits - 2); /* 16 bits precision */
    if (dx == 0) {
        put_bits(pb, 1, 0);
        put_bits(pb, 1, 1);
        put_bits(pb, nbits, dy & mask);
    } else if (dy == 0) {
        put_bits(pb, 1, 0);
        put_bits(pb, 1, 0);
        put_bits(pb, nbits, dx & mask);
    } else {
        put_bits(pb, 1, 1);
        put_bits(pb, nbits, dx & mask);
        put_bits(pb, nbits, dy & mask);
    }
}
开发者ID:Dahlqvist,项目名称:sfeMovie,代码行数:26,代码来源:swfenc.c


示例11: put_swf_matrix

static void put_swf_matrix(AVIOContext *pb,
                           int a, int b, int c, int d, int tx, int ty)
{
    PutBitContext p;
    uint8_t buf[256];
    int nbits;

    init_put_bits(&p, buf, sizeof(buf));

    put_bits(&p, 1, 1); /* a, d present */
    nbits = 1;
    max_nbits(&nbits, a);
    max_nbits(&nbits, d);
    put_bits(&p, 5, nbits); /* nb bits */
    put_bits(&p, nbits, a);
    put_bits(&p, nbits, d);

    put_bits(&p, 1, 1); /* b, c present */
    nbits = 1;
    max_nbits(&nbits, c);
    max_nbits(&nbits, b);
    put_bits(&p, 5, nbits); /* nb bits */
    put_bits(&p, nbits, c);
    put_bits(&p, nbits, b);

    nbits = 1;
    max_nbits(&nbits, tx);
    max_nbits(&nbits, ty);
    put_bits(&p, 5, nbits); /* nb bits */
    put_bits(&p, nbits, tx);
    put_bits(&p, nbits, ty);

    flush_put_bits(&p);
    avio_write(pb, buf, put_bits_ptr(&p) - p.buf);
}
开发者ID:Dahlqvist,项目名称:sfeMovie,代码行数:35,代码来源:swfenc.c


示例12: encode_window_bands_info

/**
 * Encode band info for single window group bands.
 */
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
                                     int win, int group_len, const float lambda)
{
    BandCodingPath path[120][CB_TOT_ALL];
    int w, swb, cb, start, size;
    int i, j;
    const int max_sfb  = sce->ics.max_sfb;
    const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
    const int run_esc  = (1 << run_bits) - 1;
    int idx, ppos, count;
    int stackrun[120], stackcb[120], stack_len;
    float next_minrd = INFINITY;
    int next_mincb = 0;

    abs_pow34_v(s->scoefs, sce->coeffs, 1024);
    start = win*128;
    for (cb = 0; cb < CB_TOT_ALL; cb++) {
        path[0][cb].cost     = 0.0f;
        path[0][cb].prev_idx = -1;
        path[0][cb].run      = 0;
    }
    for (swb = 0; swb < max_sfb; swb++) {
        size = sce->ics.swb_sizes[swb];
        if (sce->zeroes[win*16 + swb]) {
            for (cb = 0; cb < CB_TOT_ALL; cb++) {
                path[swb+1][cb].prev_idx = cb;
                path[swb+1][cb].cost     = path[swb][cb].cost;
                path[swb+1][cb].run      = path[swb][cb].run + 1;
            }
        } else {
            float minrd = next_minrd;
            int mincb = next_mincb;
            next_minrd = INFINITY;
            next_mincb = 0;
            for (cb = 0; cb < CB_TOT_ALL; cb++) {
                float cost_stay_here, cost_get_here;
                float rd = 0.0f;
                if (cb >= 12 && sce->band_type[win*16+swb] < aac_cb_out_map[cb] ||
                    cb  < aac_cb_in_map[sce->band_type[win*16+swb]] && sce->band_type[win*16+swb] > aac_cb_out_map[cb]) {
                    path[swb+1][cb].prev_idx = -1;
                    path[swb+1][cb].cost     = INFINITY;
                    path[swb+1][cb].run      = path[swb][cb].run + 1;
                    continue;
                }
                for (w = 0; w < group_len; w++) {
                    FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(win+w)*16+swb];
                    rd += quantize_band_cost(s, sce->coeffs + start + w*128,
                                             s->scoefs + start + w*128, size,
                                             sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
                                             lambda / band->threshold, INFINITY, NULL);
                }
                cost_stay_here = path[swb][cb].cost + rd;
                cost_get_here  = minrd              + rd + run_bits + 4;
                if (   run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
                    != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
                    cost_stay_here += run_bits;
                if (cost_get_here < cost_stay_here) {
                    path[swb+1][cb].prev_idx = mincb;
                    path[swb+1][cb].cost     = cost_get_here;
                    path[swb+1][cb].run      = 1;
                } else {
                    path[swb+1][cb].prev_idx = cb;
                    path[swb+1][cb].cost     = cost_stay_here;
                    path[swb+1][cb].run      = path[swb][cb].run + 1;
                }
                if (path[swb+1][cb].cost < next_minrd) {
                    next_minrd = path[swb+1][cb].cost;
                    next_mincb = cb;
                }
            }
        }
        start += sce->ics.swb_sizes[swb];
    }

    //convert resulting path from backward-linked list
    stack_len = 0;
    idx       = 0;
    for (cb = 1; cb < CB_TOT_ALL; cb++)
        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
            idx = cb;
    ppos = max_sfb;
    while (ppos > 0) {
        av_assert1(idx >= 0);
        cb = idx;
        stackrun[stack_len] = path[ppos][cb].run;
        stackcb [stack_len] = cb;
        idx = path[ppos-path[ppos][cb].run+1][cb].prev_idx;
        ppos -= path[ppos][cb].run;
        stack_len++;
    }
    //perform actual band info encoding
    start = 0;
    for (i = stack_len - 1; i >= 0; i--) {
        cb = aac_cb_out_map[stackcb[i]];
        put_bits(&s->pb, 4, cb);
        count = stackrun[i];
        memset(sce->zeroes + win*16 + start, !cb, count);
//.........这里部分代码省略.........
开发者ID:amersons,项目名称:FFmpeg,代码行数:101,代码来源:aaccoder.c


示例13: asv2_put_bits

static inline void asv2_put_bits(PutBitContext *pb, int n, int v){
    put_bits(pb, n, reverse[ v << (8-n) ]);
}
开发者ID:pcercuei,项目名称:dcplaya,代码行数:3,代码来源:asv1.c


示例14: put_subframe

static void put_subframe(DCAContext *c,
                         int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
                         int subframe)
{
    int i, sub, ss, ch, max_value;
    int32_t *lfe_data = (int32_t*)(c->lfe_data + 4 * SUBSUBFRAMES * subframe);

    /* Subsubframes count */
    put_bits(&c->pb, 2, SUBSUBFRAMES -1);

    /* Partial subsubframe sample count: dummy */
    put_bits(&c->pb, 3, 0);

    /* Prediction mode: no ADPCM, in each channel and subband */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++)
            put_bits(&c->pb, 1, 0);

    /* Prediction VQ addres: not transmitted */
    /* Bit allocation index */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++)
            put_bits(&c->pb, 5, QUANTIZER_BITS+3);

    if (SUBSUBFRAMES > 1) {
        /* Transition mode: none for each channel and subband */
        for (ch = 0; ch < c->prim_channels; ch++)
            for (sub = 0; sub < DCA_SUBBANDS; sub++)
                put_bits(&c->pb, 1, 0); /* codebook A4 */
    }

    /* Determine scale_factor */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++) {
            max_value = 0;
            for (i = 0; i < 8 * SUBSUBFRAMES; i++)
                max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
            c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
        }

    if (c->lfe_channel) {
        max_value = 0;
        for (i = 0; i < 4 * SUBSUBFRAMES; i++)
            max_value = FFMAX(max_value, FFABS(lfe_data[i]));
        c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
    }

    /* Scale factors: the same for each channel and subband,
       encoded according to Table D.1.2 */
    for (ch = 0; ch < c->prim_channels; ch++)
        for (sub = 0; sub < DCA_SUBBANDS; sub++)
            put_bits(&c->pb, 7, c->scale_factor[ch][sub]);

    /* Joint subband scale factor codebook select: not transmitted */
    /* Scale factors for joint subband coding: not transmitted */
    /* Stereo down-mix coefficients: not transmitted */
    /* Dynamic range coefficient: not transmitted */
    /* Stde information CRC check word: not transmitted */
    /* VQ encoded high frequency subbands: not transmitted */

    /* LFE data */
    if (c->lfe_channel) {
        for (i = 0; i < 4 * SUBSUBFRAMES; i++)
            put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
        put_bits(&c->pb, 8, c->lfe_scale_factor);
    }

    /* Audio data (subsubframes) */

    for (ss = 0; ss < SUBSUBFRAMES ; ss++)
        for (ch = 0; ch < c->prim_channels; ch++)
            for (sub = 0; sub < DCA_SUBBANDS; sub++)
                for (i = 0; i < 8; i++)
                    put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);

    /* DSYNC */
    put_bits(&c->pb, 16, 0xffff);
}
开发者ID:mcodegeeks,项目名称:OpenKODE-Framework,代码行数:78,代码来源:dcaenc.cpp


示例15: aac_adtstoasc_filter

/**
 * This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
 * ADTS header and removes the ADTS header.
 */
static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
                                AVCodecContext *avctx, const char *args,
                                uint8_t  **poutbuf, int *poutbuf_size,
                                const uint8_t *buf, int      buf_size,
                                int keyframe)
{
    GetBitContext gb;
    PutBitContext pb;
    AACADTSHeaderInfo hdr;

    AACBSFContext *ctx = bsfc->priv_data;

    init_get_bits(&gb, buf, AAC_ADTS_HEADER_SIZE*8);

    *poutbuf = (uint8_t*) buf;
    *poutbuf_size = buf_size;

    if (avctx->extradata)
        if (show_bits(&gb, 12) != 0xfff)
            return 0;

    if (avpriv_aac_parse_header(&gb, &hdr) < 0) {
        av_log(avctx, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
        return AVERROR_INVALIDDATA;
    }

    if (!hdr.crc_absent && hdr.num_aac_frames > 1) {
        avpriv_report_missing_feature(avctx,
                                      "Multiple RDBs per frame with CRC");
        return AVERROR_PATCHWELCOME;
    }

    buf      += AAC_ADTS_HEADER_SIZE + 2*!hdr.crc_absent;
    buf_size -= AAC_ADTS_HEADER_SIZE + 2*!hdr.crc_absent;

    if (!ctx->first_frame_done) {
        int            pce_size = 0;
        uint8_t        pce_data[MAX_PCE_SIZE];
        if (!hdr.chan_config) {
            init_get_bits(&gb, buf, buf_size * 8);
            if (get_bits(&gb, 3) != 5) {
                avpriv_report_missing_feature(avctx,
                                              "PCE-based channel configuration "
                                              "without PCE as first syntax "
                                              "element");
                return AVERROR_PATCHWELCOME;
            }
            init_put_bits(&pb, pce_data, MAX_PCE_SIZE);
            pce_size = avpriv_copy_pce_data(&pb, &gb)/8;
            flush_put_bits(&pb);
            buf_size -= get_bits_count(&gb)/8;
            buf      += get_bits_count(&gb)/8;
        }
        avctx->extradata_size = 2 + pce_size;
        avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);

        init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
        put_bits(&pb, 5, hdr.object_type);
        put_bits(&pb, 4, hdr.sampling_index);
        put_bits(&pb, 4, hdr.chan_config);
        put_bits(&pb, 1, 0); //frame length - 1024 samples
        put_bits(&pb, 1, 0); //does not depend on core coder
        put_bits(&pb, 1, 0); //is not extension
        flush_put_bits(&pb);
        if (pce_size) {
            memcpy(avctx->extradata + 2, pce_data, pce_size);
        }

        ctx->first_frame_done = 1;
    }

    *poutbuf = (uint8_t*) buf;
    *poutbuf_size = buf_size;

    return 0;
}
开发者ID:VFR-maniac,项目名称:libav,代码行数:80,代码来源:aac_adtstoasc_bsf.c


示例16: put_frame_header

static void put_frame_header(DCAContext *c)
{
    /* SYNC */
    put_bits(&c->pb, 16, 0x7ffe);
    put_bits(&c->pb, 16, 0x8001);

    /* Frame type: normal */
    put_bits(&c->pb, 1, 1);

    /* Deficit sample count: none */
    put_bits(&c->pb, 5, 31);

    /* CRC is not present */
    put_bits(&c->pb, 1, 0);

    /* Number of PCM sample blocks */
    put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);

    /* Primary frame byte size */
    put_bits(&c->pb, 14, c->frame_size - 1);

    /* Audio channel arrangement */
    put_bits(&c->pb, 6, c->channel_config);

    /* Core audio sampling frequency */
    put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);

    /* Transmission bit rate */
    put_bits(&c->pb, 5, c->bitrate_index);

    /* Embedded down mix: disabled */
    put_bits(&c->pb, 1, 0);

    /* Embedded dynamic range flag: not present */
    put_bits(&c->pb, 1, 0);

    /* Embedded time stamp flag: not present */
    put_bits(&c->pb, 1, 0);

    /* Auxiliary data flag: not present */
    put_bits(&c->pb, 1, 0);

    /* HDCD source: no */
    put_bits(&c->pb, 1, 0);

    /* Extension audio ID: N/A */
    put_bits(&c->pb, 3, 0);

    /* Extended audio data: not present */
    put_bits(&c->pb, 1, 0);

    /* Audio sync word insertion flag: after each sub-frame */
    put_bits(&c->pb, 1, 0);

    /* Low frequency effects flag: not present or 64x subsampling */
    put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);

    /* Predictor history switch flag: on */
    put_bits(&c->pb, 1, 1);

    /* No CRC */
    /* Multirate interpolator switch: non-perfect reconstruction */
    put_bits(&c->pb, 1, 0);

    /* Encoder software revision: 7 */
    put_bits(&c->pb, 4, 7);

    /* Copy history: 0 */
    put_bits(&c->pb, 2, 0);

    /* Source PCM resolution: 16 bits, not DTS ES */
    put_bits(&c->pb, 3, 0);

    /* Front sum/difference coding: no */
    put_bits(&c->pb, 1, 0);

    /* Surrounds sum/difference coding: no */
    put_bits(&c->pb, 1, 0);

    /* Dialog normalization: 0 dB */
    put_bits(&c->pb, 4, 0);
}
开发者ID:markjreed,项目名称:vice-emu,代码行数:82,代码来源:dcaenc.c


示例17: quantize_and_encode_band_cost_template

/**
 * Calculate rate distortion cost for quantizing with given codebook
 *
 * @return quantization distortion
 */
static av_always_inline float quantize_and_encode_band_cost_template(
                                struct AACEncContext *s,
                                PutBitContext *pb, const float *in,
                                const float *scaled, int size, int scale_idx,
                                int cb, const float lambda, const float uplim,
                                int *bits, int BT_ZERO, int BT_UNSIGNED,
                 

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