mpegaudioenc.c File Reference

#include "avcodec.h"
#include "bitstream.h"
#include "mpegaudio.h"
#include "mpegaudiodata.h"
#include "mpegaudiotab.h"

Go to the source code of this file.

Data Structures
struct	MpegAudioContext
Defines
#define	MUL(a, b)   (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
#define	SAMPLES_BUF_SIZE   4096
#define	P   15
#define	WSHIFT   (WFRAC_BITS + 15 - FRAC_BITS)
#define	SB_NOTALLOCATED   0
#define	SB_ALLOCATED   1
#define	SB_NOMORE   2
Functions
static av_cold int	MPA_encode_init (AVCodecContext *avctx)
static void	idct32 (int *out, int *tab)
static void	filter (MpegAudioContext *s, int ch, short *samples, int incr)
static void	compute_scale_factors (unsigned char scale_code[SBLIMIT], unsigned char scale_factors[SBLIMIT][3], int sb_samples[3][12][SBLIMIT], int sblimit)
static void	psycho_acoustic_model (MpegAudioContext *s, short smr[SBLIMIT])
static void	compute_bit_allocation (MpegAudioContext *s, short smr1[MPA_MAX_CHANNELS][SBLIMIT], unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], int *padding)
static void	encode_frame (MpegAudioContext *s, unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], int padding)
static int	MPA_encode_frame (AVCodecContext *avctx, unsigned char *frame, int buf_size, void *data)
static av_cold int	MPA_encode_close (AVCodecContext *avctx)
Variables
AVCodec	mp2_encoder

---

Define Documentation

#define MUL	(	a,
		b		)	(((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)

Definition at line 33 of file mpegaudioenc.c.

Referenced by idct32().

#define P   15

Referenced by direct_search(), encode_q_branch(), ff_estimate_motion_b(), ff_estimate_p_frame_motion(), ff_pre_estimate_p_frame_motion(), h263_mv4_search(), interlaced_search(), ipvideo_decode_block_opcode_0x8(), ipvideo_decode_block_opcode_0x9(), ipvideo_decode_block_opcode_0xA(), ipvideo_decode_block_opcode_0xD(), and sha1_process().

#define SAMPLES_BUF_SIZE   4096

Definition at line 35 of file mpegaudioenc.c.

Referenced by filter().

#define SB_ALLOCATED   1

Definition at line 499 of file mpegaudioenc.c.

Referenced by compute_bit_allocation().

#define SB_NOMORE   2

Definition at line 500 of file mpegaudioenc.c.

Referenced by compute_bit_allocation().

#define SB_NOTALLOCATED   0

Definition at line 498 of file mpegaudioenc.c.

Referenced by compute_bit_allocation().

#define WSHIFT   (WFRAC_BITS + 15 - FRAC_BITS)

Definition at line 310 of file mpegaudioenc.c.

Referenced by filter().

---

Function Documentation

static void compute_bit_allocation	(	MpegAudioContext *	s,
		short	smr1[MPA_MAX_CHANNELS][SBLIMIT],
		unsigned char	bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
		int *	padding
	)			[static]

Definition at line 505 of file mpegaudioenc.c.

References MpegAudioContext::alloc_table, bit_alloc(), MpegAudioContext::do_padding, MpegAudioContext::frame_frac, MpegAudioContext::frame_frac_incr, MpegAudioContext::frame_size, MPA_MAX_CHANNELS, MpegAudioContext::nb_channels, nb_scale_factors, printf, quant_snr, SB_ALLOCATED, SB_NOMORE, SB_NOTALLOCATED, MpegAudioContext::sblimit, MpegAudioContext::scale_code, and total_quant_bits.

{
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
    int incr;
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
    const unsigned char *alloc;

    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);

    /* compute frame size and padding */
    max_frame_size = s->frame_size;
    s->frame_frac += s->frame_frac_incr;
    if (s->frame_frac >= 65536) {
        s->frame_frac -= 65536;
        s->do_padding = 1;
        max_frame_size += 8;
    } else {
        s->do_padding = 0;
    }

    /* compute the header + bit alloc size */
    current_frame_size = 32;
    alloc = s->alloc_table;
    for(i=0;i<s->sblimit;i++) {
        incr = alloc[0];
        current_frame_size += incr * s->nb_channels;
        alloc += 1 << incr;
    }
    for(;;) {
        /* look for the subband with the largest signal to mask ratio */
        max_sb = -1;
        max_ch = -1;
        max_smr = INT_MIN;
        for(ch=0;ch<s->nb_channels;ch++) {
            for(i=0;i<s->sblimit;i++) {
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
                    max_smr = smr[ch][i];
                    max_sb = i;
                    max_ch = ch;
                }
            }
        }
#if 0
        printf("current=%d max=%d max_sb=%d alloc=%d\n",
               current_frame_size, max_frame_size, max_sb,
               bit_alloc[max_sb]);
#endif
        if (max_sb < 0)
            break;

        /* find alloc table entry (XXX: not optimal, should use
           pointer table) */
        alloc = s->alloc_table;
        for(i=0;i<max_sb;i++) {
            alloc += 1 << alloc[0];
        }

        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
            /* nothing was coded for this band: add the necessary bits */
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
            incr += total_quant_bits[alloc[1]];
        } else {
            /* increments bit allocation */
            b = bit_alloc[max_ch][max_sb];
            incr = total_quant_bits[alloc[b + 1]] -
                total_quant_bits[alloc[b]];
        }

        if (current_frame_size + incr <= max_frame_size) {
            /* can increase size */
            b = ++bit_alloc[max_ch][max_sb];
            current_frame_size += incr;
            /* decrease smr by the resolution we added */
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
            /* max allocation size reached ? */
            if (b == ((1 << alloc[0]) - 1))
                subband_status[max_ch][max_sb] = SB_NOMORE;
            else
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
        } else {
            /* cannot increase the size of this subband */
            subband_status[max_ch][max_sb] = SB_NOMORE;
        }
    }
    *padding = max_frame_size - current_frame_size;
    assert(*padding >= 0);

#if 0
    for(i=0;i<s->sblimit;i++) {
        printf("%d ", bit_alloc[i]);
    }
    printf("\n");
#endif
}

static void compute_scale_factors	(	unsigned char	scale_code[SBLIMIT],
		unsigned char	scale_factors[SBLIMIT][3],
		int	sb_samples[3][12][SBLIMIT],
		int	sblimit
	)			[static]

Definition at line 369 of file mpegaudioenc.c.

References av_log2(), printf, scale_diff_table, and scale_factor_table.

Referenced by MPA_encode_frame().

{
    int *p, vmax, v, n, i, j, k, code;
    int index, d1, d2;
    unsigned char *sf = &scale_factors[0][0];

    for(j=0;j<sblimit;j++) {
        for(i=0;i<3;i++) {
            /* find the max absolute value */
            p = &sb_samples[i][0][j];
            vmax = abs(*p);
            for(k=1;k<12;k++) {
                p += SBLIMIT;
                v = abs(*p);
                if (v > vmax)
                    vmax = v;
            }
            /* compute the scale factor index using log 2 computations */
            if (vmax > 1) {
                n = av_log2(vmax);
                /* n is the position of the MSB of vmax. now
                   use at most 2 compares to find the index */
                index = (21 - n) * 3 - 3;
                if (index >= 0) {
                    while (vmax <= scale_factor_table[index+1])
                        index++;
                } else {
                    index = 0; /* very unlikely case of overflow */
                }
            } else {
                index = 62; /* value 63 is not allowed */
            }

#if 0
            printf("%2d:%d in=%x %x %d\n",
                   j, i, vmax, scale_factor_table[index], index);
#endif
            /* store the scale factor */
            assert(index >=0 && index <= 63);
            sf[i] = index;
        }

        /* compute the transmission factor : look if the scale factors
           are close enough to each other */
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
        d2 = scale_diff_table[sf[1] - sf[2] + 64];

        /* handle the 25 cases */
        switch(d1 * 5 + d2) {
        case 0*5+0:
        case 0*5+4:
        case 3*5+4:
        case 4*5+0:
        case 4*5+4:
            code = 0;
            break;
        case 0*5+1:
        case 0*5+2:
        case 4*5+1:
        case 4*5+2:
            code = 3;
            sf[2] = sf[1];
            break;
        case 0*5+3:
        case 4*5+3:
            code = 3;
            sf[1] = sf[2];
            break;
        case 1*5+0:
        case 1*5+4:
        case 2*5+4:
            code = 1;
            sf[1] = sf[0];
            break;
        case 1*5+1:
        case 1*5+2:
        case 2*5+0:
        case 2*5+1:
        case 2*5+2:
            code = 2;
            sf[1] = sf[2] = sf[0];
            break;
        case 2*5+3:
        case 3*5+3:
            code = 2;
            sf[0] = sf[1] = sf[2];
            break;
        case 3*5+0:
        case 3*5+1:
        case 3*5+2:
            code = 2;
            sf[0] = sf[2] = sf[1];
            break;
        case 1*5+3:
            code = 2;
            if (sf[0] > sf[2])
              sf[0] = sf[2];
            sf[1] = sf[2] = sf[0];
            break;
        default:
            assert(0); //cannot happen
            code = 0;           /* kill warning */
        }

#if 0
        printf("%d: %2d %2d %2d %d %d -> %d\n", j,
               sf[0], sf[1], sf[2], d1, d2, code);
#endif
        scale_code[j] = code;
        sf += 3;
    }
}

static void encode_frame	(	MpegAudioContext *	s,
		unsigned char	bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
		int	padding
	)			[static]

Definition at line 610 of file mpegaudioenc.c.

References MpegAudioContext::alloc_table, bit_alloc(), MpegAudioContext::bitrate_index, MpegAudioContext::do_padding, MpegAudioContext::freq_index, MpegAudioContext::lsf, MPA_MONO, MPA_STEREO, MpegAudioContext::nb_channels, MpegAudioContext::pb, put_bits(), and MpegAudioContext::sblimit.

{
    int i, j, k, l, bit_alloc_bits, b, ch;
    unsigned char *sf;
    int q[3];
    PutBitContext *p = &s->pb;

    /* header */

    put_bits(p, 12, 0xfff);
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
    put_bits(p, 2, 4-2);  /* layer 2 */
    put_bits(p, 1, 1); /* no error protection */
    put_bits(p, 4, s->bitrate_index);
    put_bits(p, 2, s->freq_index);
    put_bits(p, 1, s->do_padding); /* use padding */
    put_bits(p, 1, 0);             /* private_bit */
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
    put_bits(p, 2, 0); /* mode_ext */
    put_bits(p, 1, 0); /* no copyright */
    put_bits(p, 1, 1); /* original */
    put_bits(p, 2, 0); /* no emphasis */

    /* bit allocation */
    j = 0;
    for(i=0;i<s->sblimit;i++) {
        bit_alloc_bits = s->alloc_table[j];
        for(ch=0;ch<s->nb_channels;ch++) {
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
        }
        j += 1 << bit_alloc_bits;
    }

    /* scale codes */
    for(i=0;i<s->sblimit;i++) {
        for(ch=0;ch<s->nb_channels;ch++) {
            if (bit_alloc[ch][i])
                put_bits(p, 2, s->scale_code[ch][i]);
        }
    }

    /* scale factors */
    for(i=0;i<s->sblimit;i++) {
        for(ch=0;ch<s->nb_channels;ch++) {
            if (bit_alloc[ch][i]) {
                sf = &s->scale_factors[ch][i][0];
                switch(s->scale_code[ch][i]) {
                case 0:
                    put_bits(p, 6, sf[0]);
                    put_bits(p, 6, sf[1]);
                    put_bits(p, 6, sf[2]);
                    break;
                case 3:
                case 1:
                    put_bits(p, 6, sf[0]);
                    put_bits(p, 6, sf[2]);
                    break;
                case 2:
                    put_bits(p, 6, sf[0]);
                    break;
                }
            }
        }
    }

    /* quantization & write sub band samples */

    for(k=0;k<3;k++) {
        for(l=0;l<12;l+=3) {
            j = 0;
            for(i=0;i<s->sblimit;i++) {
                bit_alloc_bits = s->alloc_table[j];
                for(ch=0;ch<s->nb_channels;ch++) {
                    b = bit_alloc[ch][i];
                    if (b) {
                        int qindex, steps, m, sample, bits;
                        /* we encode 3 sub band samples of the same sub band at a time */
                        qindex = s->alloc_table[j+b];
                        steps = ff_mpa_quant_steps[qindex];
                        for(m=0;m<3;m++) {
                            sample = s->sb_samples[ch][k][l + m][i];
                            /* divide by scale factor */
#ifdef USE_FLOATS
                            {
                                float a;
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
                                q[m] = (int)((a + 1.0) * steps * 0.5);
                            }
#else
                            {
                                int q1, e, shift, mult;
                                e = s->scale_factors[ch][i][k];
                                shift = scale_factor_shift[e];
                                mult = scale_factor_mult[e];

                                /* normalize to P bits */
                                if (shift < 0)
                                    q1 = sample << (-shift);
                                else
                                    q1 = sample >> shift;
                                q1 = (q1 * mult) >> P;
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
                            }
#endif
                            if (q[m] >= steps)
                                q[m] = steps - 1;
                            assert(q[m] >= 0 && q[m] < steps);
                        }
                        bits = ff_mpa_quant_bits[qindex];
                        if (bits < 0) {
                            /* group the 3 values to save bits */
                            put_bits(p, -bits,
                                     q[0] + steps * (q[1] + steps * q[2]));
#if 0
                            printf("%d: gr1 %d\n",
                                   i, q[0] + steps * (q[1] + steps * q[2]));
#endif
                        } else {
#if 0
                            printf("%d: gr3 %d %d %d\n",
                                   i, q[0], q[1], q[2]);
#endif
                            put_bits(p, bits, q[0]);
                            put_bits(p, bits, q[1]);
                            put_bits(p, bits, q[2]);
                        }
                    }
                }
                /* next subband in alloc table */
                j += 1 << bit_alloc_bits;
            }
        }
    }

    /* padding */
    for(i=0;i<padding;i++)
        put_bits(p, 1, 0);

    /* flush */
    flush_put_bits(p);
}

static void filter	(	MpegAudioContext *	s,
		int	ch,
		short *	samples,
		int	incr
	)			[static]

Definition at line 312 of file mpegaudioenc.c.

References filter_bank, idct32(), offset, MpegAudioContext::samples_buf, SAMPLES_BUF_SIZE, MpegAudioContext::samples_offset, MpegAudioContext::sb_samples, and WSHIFT.

Referenced by av_resample(), avfilter_config_links(), avfilter_destroy(), avfilter_graph_add_filter(), avfilter_init_filter(), avfilter_open(), avfilter_parse_graph(), avfilter_register(), filter_name(), h_resample_fast(), link_filter_inouts(), MPA_encode_frame(), mpegts_close_filter(), mpegts_open_pes_filter(), mpegts_open_section_filter(), mpegts_push_data(), pat_cb(), pick_formats(), pmt_cb(), query_formats(), sdt_cb(), truespeech_apply_twopoint_filter(), tta_decode_frame(), and xa_decode().

{
    short *p, *q;
    int sum, offset, i, j;
    int tmp[64];
    int tmp1[32];
    int *out;

    //    print_pow1(samples, 1152);

    offset = s->samples_offset[ch];
    out = &s->sb_samples[ch][0][0][0];
    for(j=0;j<36;j++) {
        /* 32 samples at once */
        for(i=0;i<32;i++) {
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
            samples += incr;
        }

        /* filter */
        p = s->samples_buf[ch] + offset;
        q = filter_bank;
        /* maxsum = 23169 */
        for(i=0;i<64;i++) {
            sum = p[0*64] * q[0*64];
            sum += p[1*64] * q[1*64];
            sum += p[2*64] * q[2*64];
            sum += p[3*64] * q[3*64];
            sum += p[4*64] * q[4*64];
            sum += p[5*64] * q[5*64];
            sum += p[6*64] * q[6*64];
            sum += p[7*64] * q[7*64];
            tmp[i] = sum;
            p++;
            q++;
        }
        tmp1[0] = tmp[16] >> WSHIFT;
        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;

        idct32(out, tmp1);

        /* advance of 32 samples */
        offset -= 32;
        out += 32;
        /* handle the wrap around */
        if (offset < 0) {
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
                    s->samples_buf[ch], (512 - 32) * 2);
            offset = SAMPLES_BUF_SIZE - 512;
        }
    }
    s->samples_offset[ch] = offset;

    //    print_pow(s->sb_samples, 1152);
}

static void idct32	(	int *	out,
		int *	tab
	)			[static]

Definition at line 191 of file mpegaudioenc.c.

References bitinv32, costab32, FIX, MUL, SQRT2, t, and t1.

Referenced by filter().

{
    int i, j;
    int *t, *t1, xr;
    const int *xp = costab32;

    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];

    t = tab + 30;
    t1 = tab + 2;
    do {
        t[0] += t[-4];
        t[1] += t[1 - 4];
        t -= 4;
    } while (t != t1);

    t = tab + 28;
    t1 = tab + 4;
    do {
        t[0] += t[-8];
        t[1] += t[1-8];
        t[2] += t[2-8];
        t[3] += t[3-8];
        t -= 8;
    } while (t != t1);

    t = tab;
    t1 = tab + 32;
    do {
        t[ 3] = -t[ 3];
        t[ 6] = -t[ 6];

        t[11] = -t[11];
        t[12] = -t[12];
        t[13] = -t[13];
        t[15] = -t[15];
        t += 16;
    } while (t != t1);


    t = tab;
    t1 = tab + 8;
    do {
        int x1, x2, x3, x4;

        x3 = MUL(t[16], FIX(SQRT2*0.5));
        x4 = t[0] - x3;
        x3 = t[0] + x3;

        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
        x1 = MUL((t[8] - x2), xp[0]);
        x2 = MUL((t[8] + x2), xp[1]);

        t[ 0] = x3 + x1;
        t[ 8] = x4 - x2;
        t[16] = x4 + x2;
        t[24] = x3 - x1;
        t++;
    } while (t != t1);

    xp += 2;
    t = tab;
    t1 = tab + 4;
    do {
        xr = MUL(t[28],xp[0]);
        t[28] = (t[0] - xr);
        t[0] = (t[0] + xr);

        xr = MUL(t[4],xp[1]);
        t[ 4] = (t[24] - xr);
        t[24] = (t[24] + xr);

        xr = MUL(t[20],xp[2]);
        t[20] = (t[8] - xr);
        t[ 8] = (t[8] + xr);

        xr = MUL(t[12],xp[3]);
        t[12] = (t[16] - xr);
        t[16] = (t[16] + xr);
        t++;
    } while (t != t1);
    xp += 4;

    for (i = 0; i < 4; i++) {
        xr = MUL(tab[30-i*4],xp[0]);
        tab[30-i*4] = (tab[i*4] - xr);
        tab[   i*4] = (tab[i*4] + xr);

        xr = MUL(tab[ 2+i*4],xp[1]);
        tab[ 2+i*4] = (tab[28-i*4] - xr);
        tab[28-i*4] = (tab[28-i*4] + xr);

        xr = MUL(tab[31-i*4],xp[0]);
        tab[31-i*4] = (tab[1+i*4] - xr);
        tab[ 1+i*4] = (tab[1+i*4] + xr);

        xr = MUL(tab[ 3+i*4],xp[1]);
        tab[ 3+i*4] = (tab[29-i*4] - xr);
        tab[29-i*4] = (tab[29-i*4] + xr);

        xp += 2;
    }

    t = tab + 30;
    t1 = tab + 1;
    do {
        xr = MUL(t1[0], *xp);
        t1[0] = (t[0] - xr);
        t[0] = (t[0] + xr);
        t -= 2;
        t1 += 2;
        xp++;
    } while (t >= tab);

    for(i=0;i<32;i++) {
        out[i] = tab[bitinv32[i]];
    }
}

static av_cold int MPA_encode_close

(

AVCodecContext *

avctx

)

[static]

Definition at line 784 of file mpegaudioenc.c.

References av_freep(), and AVCodecContext::coded_frame.

{
    av_freep(&avctx->coded_frame);
    return 0;
}

static int MPA_encode_frame	(	AVCodecContext *	avctx,
		unsigned char *	frame,
		int	buf_size,
		void *	data
	)			[static]

Definition at line 754 of file mpegaudioenc.c.

References bit_alloc(), compute_bit_allocation(), compute_scale_factors(), encode_frame(), filter(), init_put_bits(), MPA_FRAME_SIZE, MPA_MAX_CHANNELS, MPA_MAX_CODED_FRAME_SIZE, MpegAudioContext::nb_channels, pbBufPtr(), AVCodecContext::priv_data, psycho_acoustic_model(), samples, MpegAudioContext::sb_samples, MpegAudioContext::sblimit, SBLIMIT, MpegAudioContext::scale_code, and MpegAudioContext::scale_factors.

{
    MpegAudioContext *s = avctx->priv_data;
    short *samples = data;
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
    int padding, i;

    for(i=0;i<s->nb_channels;i++) {
        filter(s, i, samples + i, s->nb_channels);
    }

    for(i=0;i<s->nb_channels;i++) {
        compute_scale_factors(s->scale_code[i], s->scale_factors[i],
                              s->sb_samples[i], s->sblimit);
    }
    for(i=0;i<s->nb_channels;i++) {
        psycho_acoustic_model(s, smr[i]);
    }
    compute_bit_allocation(s, smr, bit_alloc, &padding);

    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);

    encode_frame(s, bit_alloc, padding);

    s->nb_samples += MPA_FRAME_SIZE;
    return pbBufPtr(&s->pb) - s->pb.buf;
}

static av_cold int MPA_encode_init

(

AVCodecContext *

avctx

)

[static]

Definition at line 64 of file mpegaudioenc.c.

References MpegAudioContext::alloc_table, av_log(), AV_LOG_DEBUG, AV_LOG_ERROR, MpegAudioContext::bit_rate, AVCodecContext::bit_rate, MpegAudioContext::bitrate_index, AVCodecContext::channels, ff_mpa_alloc_tables, ff_mpa_bitrate_tab, ff_mpa_enwindow, ff_mpa_freq_tab, ff_mpa_l2_select_table(), ff_mpa_sblimit_table, filter_bank, MpegAudioContext::frame_frac, MpegAudioContext::frame_frac_incr, MpegAudioContext::frame_size, AVCodecContext::frame_size, MpegAudioContext::freq, MpegAudioContext::freq_index, MpegAudioContext::lsf, MPA_FRAME_SIZE, MpegAudioContext::nb_channels, AVCodecContext::priv_data, AVCodecContext::sample_rate, MpegAudioContext::samples_offset, MpegAudioContext::sblimit, and WFRAC_BITS.

{
    MpegAudioContext *s = avctx->priv_data;
    int freq = avctx->sample_rate;
    int bitrate = avctx->bit_rate;
    int channels = avctx->channels;
    int i, v, table;
    float a;

    if (channels <= 0 || channels > 2){
        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
        return -1;
    }
    bitrate = bitrate / 1000;
    s->nb_channels = channels;
    s->freq = freq;
    s->bit_rate = bitrate * 1000;
    avctx->frame_size = MPA_FRAME_SIZE;

    /* encoding freq */
    s->lsf = 0;
    for(i=0;i<3;i++) {
        if (ff_mpa_freq_tab[i] == freq)
            break;
        if ((ff_mpa_freq_tab[i] / 2) == freq) {
            s->lsf = 1;
            break;
        }
    }
    if (i == 3){
        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
        return -1;
    }
    s->freq_index = i;

    /* encoding bitrate & frequency */
    for(i=0;i<15;i++) {
        if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
            break;
    }
    if (i == 15){
        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
        return -1;
    }
    s->bitrate_index = i;

    /* compute total header size & pad bit */

    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
    s->frame_size = ((int)a) * 8;

    /* frame fractional size to compute padding */
    s->frame_frac = 0;
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);

    /* select the right allocation table */
    table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);

    /* number of used subbands */
    s->sblimit = ff_mpa_sblimit_table[table];
    s->alloc_table = ff_mpa_alloc_tables[table];

#ifdef DEBUG
    av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
#endif

    for(i=0;i<s->nb_channels;i++)
        s->samples_offset[i] = 0;

    for(i=0;i<257;i++) {
        int v;
        v = ff_mpa_enwindow[i];
#if WFRAC_BITS != 16
        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif
        filter_bank[i] = v;
        if ((i & 63) != 0)
            v = -v;
        if (i != 0)
            filter_bank[512 - i] = v;
    }

    for(i=0;i<64;i++) {
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
        if (v <= 0)
            v = 1;
        scale_factor_table[i] = v;
#ifdef USE_FLOATS
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
#else
#define P 15
        scale_factor_shift[i] = 21 - P - (i / 3);
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
#endif
    }
    for(i=0;i<128;i++) {
        v = i - 64;
        if (v <= -3)
            v = 0;
        else if (v < 0)
            v = 1;
        else if (v == 0)
            v = 2;
        else if (v < 3)
            v = 3;
        else
            v = 4;
        scale_diff_table[i] = v;
    }

    for(i=0;i<17;i++) {
        v = ff_mpa_quant_bits[i];
        if (v < 0)
            v = -v;
        else
            v = v * 3;
        total_quant_bits[i] = 12 * v;
    }

    avctx->coded_frame= avcodec_alloc_frame();
    avctx->coded_frame->key_frame= 1;

    return 0;
}

static void psycho_acoustic_model	(	MpegAudioContext *	s,
		short	smr[SBLIMIT]
	)			[static]

Definition at line 488 of file mpegaudioenc.c.

References fixed_smr, and MpegAudioContext::sblimit.

Referenced by MPA_encode_frame().

{
    int i;

    for(i=0;i<s->sblimit;i++) {
        smr[i] = (int)(fixed_smr[i] * 10);
    }
}

---

Variable Documentation

AVCodec mp2_encoder

Initial value:

 {
    "mp2",
    CODEC_TYPE_AUDIO,
    CODEC_ID_MP2,
    sizeof(MpegAudioContext),
    MPA_encode_init,
    MPA_encode_frame,
    MPA_encode_close,
    NULL,
    .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
}

Definition at line 790 of file mpegaudioenc.c.

---