comparison vorbis_dec.c @ 7261:032a49f033e8 libavcodec

simplify vorbis windowing

7260:3ec34b551aae

    uint_fast8_t mapping_count;
    vorbis_mapping *mappings;
    uint_fast8_t mode_count;
    vorbis_mode *modes;
    uint_fast8_t mode_number; // mode number for the current packet
    float *channel_residues;
    float *channel_floors;
    float *saved;
    uint_fast16_t saved_start;
    float *ret;
    float *buf;
    float *buf_tmp;
    uint_fast32_t add_bias; // for float->int conversion
    uint_fast32_t exp_bias;

    vc->channel_floors  = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->saved           = av_mallocz((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->ret             = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->buf             = av_malloc( vc->blocksize[1]                       * sizeof(float));
    vc->buf_tmp         = av_malloc( vc->blocksize[1]                       * sizeof(float));
    vc->saved_start=0;

    ff_mdct_init(&vc->mdct[0], bl0, 1);
    ff_mdct_init(&vc->mdct[1], bl1, 1);

    AV_DEBUG(" vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",

            }
        }
    }
}

// Decode the audio packet using the functions above

static int vorbis_parse_audio_packet(vorbis_context *vc) {
    GetBitContext *gb=&vc->gb;

    uint_fast8_t previous_window=0,next_window=0;
    uint_fast8_t mode_number;
    uint_fast16_t blocksize;
    int_fast32_t i,j;
    uint_fast8_t no_residue[vc->audio_channels];
    uint_fast8_t do_not_decode[vc->audio_channels];
    vorbis_mapping *mapping;
    float *ch_res_ptr=vc->channel_residues;
    float *ch_floor_ptr=vc->channel_floors;
    uint_fast8_t res_chan[vc->audio_channels];
    uint_fast8_t res_num=0;
    int_fast16_t retlen=0;
    uint_fast16_t saved_start=0;
    float fadd_bias = vc->add_bias;

    if (get_bits1(gb)) {
        av_log(vc->avccontext, AV_LOG_ERROR, "Not a Vorbis I audio packet.\n");
        return -1; // packet type not audio

    vc->mode_number=mode_number;
    mapping=&vc->mappings[vc->modes[mode_number].mapping];

    AV_DEBUG(" Mode number: %d , mapping: %d , blocktype %d \n", mode_number, vc->modes[mode_number].mapping, vc->modes[mode_number].blockflag);

    if (vc->modes[mode_number].blockflag) {
        previous_window=get_bits1(gb);
        next_window=get_bits1(gb);
    }

    blocksize=vc->blocksize[vc->modes[mode_number].blockflag];
    memset(ch_res_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
    memset(ch_floor_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?

// Decode floor

        vc->dsp.vector_fmul(ch_floor_ptr, ch_res_ptr, blocksize/2);
    }

// MDCT, overlap/add, save data for next overlapping  FPMATH

    for(j=0;j<vc->audio_channels;++j) {
        uint_fast8_t step=vc->audio_channels;
        uint_fast16_t k;
        float *saved=vc->saved+j*vc->blocksize[1]/2;
        float *ret=vc->ret;
        const float *lwin=vc->win[1];
        const float *swin=vc->win[0];
        float *buf=vc->buf;
        float *buf_tmp=vc->buf_tmp;

        ch_floor_ptr=vc->channel_floors+j*blocksize/2;

        saved_start=vc->saved_start;

        vc->mdct[0].fft.imdct_calc(&vc->mdct[vc->modes[mode_number].blockflag], buf, ch_floor_ptr, buf_tmp);

        //FIXME process channels together, to allow faster simd vector_fmul_add_add?
        if (vc->modes[mode_number].blockflag) {
            // -- overlap/add
            if (previous_window) {
                vc->dsp.vector_fmul_add_add(ret+j, buf, lwin, saved, vc->add_bias, vc->blocksize[1]/2, step);
                retlen=vc->blocksize[1]/2;
            } else {
                int len = (vc->blocksize[1]-vc->blocksize[0])/4;
                buf += len;
                vc->dsp.vector_fmul_add_add(ret+j, buf, swin, saved, vc->add_bias, vc->blocksize[0]/2, step);
                k = vc->blocksize[0]/2*step + j;
                buf += vc->blocksize[0]/2;
                if(vc->exp_bias){
                    for(i=0; i<len; i++, k+=step)
                        ((uint32_t*)ret)[k] = ((uint32_t*)buf)[i] + vc->exp_bias; // ret[k]=buf[i]*(1<<bias)
                } else {
                    for(i=0; i<len; i++, k+=step)
                        ret[k] = buf[i] + fadd_bias;
                }
                buf=vc->buf;
                retlen=vc->blocksize[0]/2+len;
            }
            // -- save
            if (next_window) {
                buf += vc->blocksize[1]/2;
                vc->dsp.vector_fmul_reverse(saved, buf, lwin, vc->blocksize[1]/2);
                saved_start=0;
            } else {
                saved_start=(vc->blocksize[1]-vc->blocksize[0])/4;
                buf += vc->blocksize[1]/2;
                for(i=0; i<saved_start; i++)
                    ((uint32_t*)saved)[i] = ((uint32_t*)buf)[i] + vc->exp_bias;
                vc->dsp.vector_fmul_reverse(saved+saved_start, buf+saved_start, swin, vc->blocksize[0]/2);
            }
        } else {
            // --overlap/add
            if(vc->add_bias) {
                for(k=j, i=0;i<saved_start;++i, k+=step)
                    ret[k] = saved[i] + fadd_bias;
            } else {
                for(k=j, i=0;i<saved_start;++i, k+=step)
                    ret[k] = saved[i];
            }
            vc->dsp.vector_fmul_add_add(ret+k, buf, swin, saved+saved_start, vc->add_bias, vc->blocksize[0]/2, step);
            retlen=saved_start+vc->blocksize[0]/2;
            // -- save
            buf += vc->blocksize[0]/2;
            vc->dsp.vector_fmul_reverse(saved, buf, swin, vc->blocksize[0]/2);
            saved_start=0;
        }
    }
    vc->saved_start=saved_start;

    return retlen*vc->audio_channels;
}

// Return the decoded audio packet through the standard api

static int vorbis_decode_frame(AVCodecContext *avccontext,

        return buf_size ;
    }

    AV_DEBUG("parsed %d bytes %d bits, returned %d samples (*ch*bits) \n", get_bits_count(gb)/8, get_bits_count(gb)%8, len);

    vc->dsp.float_to_int16(data, vc->ret, len);
    *data_size=len*2;

    return buf_size ;
}

// Close decoder

7261:032a49f033e8

    uint_fast8_t mapping_count;
    vorbis_mapping *mappings;
    uint_fast8_t mode_count;
    vorbis_mode *modes;
    uint_fast8_t mode_number; // mode number for the current packet
    uint_fast8_t previous_window;
    float *channel_residues;
    float *channel_floors;
    float *saved;
    float *ret;
    float *buf;
    float *buf_tmp;
    uint_fast32_t add_bias; // for float->int conversion
    uint_fast32_t exp_bias;

    vc->channel_floors  = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->saved           = av_mallocz((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->ret             = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
    vc->buf             = av_malloc( vc->blocksize[1]                       * sizeof(float));
    vc->buf_tmp         = av_malloc( vc->blocksize[1]                       * sizeof(float));
    vc->previous_window=0;

    ff_mdct_init(&vc->mdct[0], bl0, 1);
    ff_mdct_init(&vc->mdct[1], bl1, 1);

    AV_DEBUG(" vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",

            }
        }
    }
}

static void copy_normalize(float *dst, float *src, int len, int exp_bias, float add_bias)
{
    int i;
    if(exp_bias) {
        for(i=0; i<len; i++)
            ((uint32_t*)dst)[i] = ((uint32_t*)src)[i] + exp_bias; // dst[k]=src[i]*(1<<bias)
    } else {
        for(i=0; i<len; i++)
            dst[i] = src[i] + add_bias;
    }
}

// Decode the audio packet using the functions above

static int vorbis_parse_audio_packet(vorbis_context *vc) {
    GetBitContext *gb=&vc->gb;

    uint_fast8_t previous_window=vc->previous_window;
    uint_fast8_t mode_number;
    uint_fast8_t blockflag;
    uint_fast16_t blocksize;
    int_fast32_t i,j;
    uint_fast8_t no_residue[vc->audio_channels];
    uint_fast8_t do_not_decode[vc->audio_channels];
    vorbis_mapping *mapping;
    float *ch_res_ptr=vc->channel_residues;
    float *ch_floor_ptr=vc->channel_floors;
    uint_fast8_t res_chan[vc->audio_channels];
    uint_fast8_t res_num=0;
    int_fast16_t retlen=0;
    float fadd_bias = vc->add_bias;

    if (get_bits1(gb)) {
        av_log(vc->avccontext, AV_LOG_ERROR, "Not a Vorbis I audio packet.\n");
        return -1; // packet type not audio

    vc->mode_number=mode_number;
    mapping=&vc->mappings[vc->modes[mode_number].mapping];

    AV_DEBUG(" Mode number: %d , mapping: %d , blocktype %d \n", mode_number, vc->modes[mode_number].mapping, vc->modes[mode_number].blockflag);

    blockflag=vc->modes[mode_number].blockflag;
    blocksize=vc->blocksize[blockflag];
    if (blockflag) {
        skip_bits(gb, 2); // previous_window, next_window
    }

    memset(ch_res_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
    memset(ch_floor_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?

// Decode floor

        vc->dsp.vector_fmul(ch_floor_ptr, ch_res_ptr, blocksize/2);
    }

// MDCT, overlap/add, save data for next overlapping  FPMATH

    retlen = (blocksize + vc->blocksize[previous_window])/4;
    for(j=0;j<vc->audio_channels;++j) {
        uint_fast16_t bs0=vc->blocksize[0];
        uint_fast16_t bs1=vc->blocksize[1];
        float *saved=vc->saved+j*bs1/2;
        float *ret=vc->ret+j*retlen;
        float *buf=vc->buf;
        const float *win=vc->win[blockflag&previous_window];

        vc->mdct[0].fft.imdct_calc(&vc->mdct[blockflag], buf, vc->channel_floors+j*blocksize/2, vc->buf_tmp);

        if(blockflag == previous_window) {
            vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, blocksize/2);
        } else if(blockflag > previous_window) {
            vc->dsp.vector_fmul_window(ret, saved, buf+(bs1-bs0)/4, win, fadd_bias, bs0/2);
            copy_normalize(ret+bs0/2, buf+(bs1+bs0)/4, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
        } else {
            copy_normalize(ret, saved, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
            vc->dsp.vector_fmul_window(ret+(bs1-bs0)/4, saved+(bs1-bs0)/4, buf, win, fadd_bias, bs0/2);
        }
        memcpy(saved, buf+blocksize/2, blocksize/2*sizeof(float));
    }

    vc->previous_window = blockflag;
    return retlen;
}

// Return the decoded audio packet through the standard api

static int vorbis_decode_frame(AVCodecContext *avccontext,

        return buf_size ;
    }

    AV_DEBUG("parsed %d bytes %d bits, returned %d samples (*ch*bits) \n", get_bits_count(gb)/8, get_bits_count(gb)%8, len);

    vc->dsp.float_to_int16_interleave(data, vc->ret, len, vc->audio_channels);
    *data_size=len*2*vc->audio_channels;

    return buf_size ;
}

// Close decoder