Mercurial > libavcodec.hg
changeset 5052:d981eb275c8f libavcodec
remove dependency of mpeg audio encoder over mpeg audio decoder
| author | aurel |
|---|---|
| date | Sun, 20 May 2007 22:50:29 +0000 |
| parents | b908c67063c8 |
| children | 34a5f37c4bf7 |
| files | Makefile mpegaudio.c mpegaudio.h mpegaudiodec.c mpegaudioenc.c |
| diffstat | 5 files changed, 826 insertions(+), 797 deletions(-) [+] |
line wrap: on
line diff
--- a/Makefile Sun May 20 15:11:55 2007 +0000 +++ b/Makefile Sun May 20 22:50:29 2007 +0000 @@ -19,7 +19,7 @@ motion_est.o \ imgconvert.o \ mpeg12.o \ - mpegaudiodec.o mpegaudiodecheader.o mpegaudiodata.o \ + mpegaudiodec.o mpegaudiodecheader.o mpegaudiodata.o mpegaudio.o \ simple_idct.o \ ratecontrol.o \ eval.o \ @@ -108,7 +108,7 @@ OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o mpegvideo.o OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o mjpegdec.o mjpeg.o OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o -OBJS-$(CONFIG_MP2_ENCODER) += mpegaudio.o mpegaudiodata.o +OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o mpegaudiodata.o OBJS-$(CONFIG_MPC7_DECODER) += mpc.o OBJS-$(CONFIG_MSMPEG4V1_DECODER) += msmpeg4.o msmpeg4data.o OBJS-$(CONFIG_MSMPEG4V1_ENCODER) += msmpeg4.o msmpeg4data.o
--- a/mpegaudio.c Sun May 20 15:11:55 2007 +0000 +++ b/mpegaudio.c Sun May 20 22:50:29 2007 +0000 @@ -1,6 +1,6 @@ /* - * The simplest mpeg audio layer 2 encoder - * Copyright (c) 2000, 2001 Fabrice Bellard. + * MPEG Audio common code + * Copyright (c) 2001, 2002 Fabrice Bellard. * * This file is part of FFmpeg. * @@ -21,782 +21,30 @@ /** * @file mpegaudio.c - * The simplest mpeg audio layer 2 encoder. + * MPEG Audio common code. */ -#include "avcodec.h" -#include "bitstream.h" #include "mpegaudio.h" -/* currently, cannot change these constants (need to modify - quantization stage) */ -#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) -#define FIX(a) ((int)((a) * (1 << FRAC_BITS))) - -#define SAMPLES_BUF_SIZE 4096 - -typedef struct MpegAudioContext { - PutBitContext pb; - int nb_channels; - int freq, bit_rate; - int lsf; /* 1 if mpeg2 low bitrate selected */ - int bitrate_index; /* bit rate */ - int freq_index; - int frame_size; /* frame size, in bits, without padding */ - int64_t nb_samples; /* total number of samples encoded */ - /* padding computation */ - int frame_frac, frame_frac_incr, do_padding; - short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ - int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ - int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; - unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ - /* code to group 3 scale factors */ - unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; - int sblimit; /* number of used subbands */ - const unsigned char *alloc_table; -} MpegAudioContext; - -/* define it to use floats in quantization (I don't like floats !) */ -//#define USE_FLOATS - -#include "mpegaudiodata.h" -#include "mpegaudiotab.h" - -static int MPA_encode_init(AVCodecContext *avctx) -{ - 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 = 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; -} - -/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ -static void idct32(int *out, int *tab) -{ - 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]]; - } -} - -#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) - -static void filter(MpegAudioContext *s, int ch, short *samples, int incr) +/* bitrate is in kb/s */ +int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf) { - 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 compute_scale_factors(unsigned char scale_code[SBLIMIT], - unsigned char scale_factors[SBLIMIT][3], - int sb_samples[3][12][SBLIMIT], - int sblimit) -{ - 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 > 0) { - 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); //cant 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; - } -} - -/* The most important function : psycho acoustic module. In this - encoder there is basically none, so this is the worst you can do, - but also this is the simpler. */ -static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) -{ - int i; - - for(i=0;i<s->sblimit;i++) { - smr[i] = (int)(fixed_smr[i] * 10); - } -} - - -#define SB_NOTALLOCATED 0 -#define SB_ALLOCATED 1 -#define SB_NOMORE 2 - -/* Try to maximize the smr while using a number of bits inferior to - the frame size. I tried to make the code simpler, faster and - smaller than other encoders :-) */ -static void compute_bit_allocation(MpegAudioContext *s, - short smr1[MPA_MAX_CHANNELS][SBLIMIT], - unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], - int *padding) -{ - 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; + int ch_bitrate, table; - 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; + ch_bitrate = bitrate / nb_channels; + if (!lsf) { + if ((freq == 48000 && ch_bitrate >= 56) || + (ch_bitrate >= 56 && ch_bitrate <= 80)) + table = 0; + else if (freq != 48000 && ch_bitrate >= 96) + table = 1; + else if (freq != 32000 && ch_bitrate <= 48) + table = 2; + else + table = 3; } 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 = 0x80000000; - 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 -} - -/* - * Output the mpeg audio layer 2 frame. Note how the code is small - * compared to other encoders :-) - */ -static void encode_frame(MpegAudioContext *s, - unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], - int padding) -{ - 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]); - } + table = 4; } - - /* 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); + return table; } - -static int MPA_encode_frame(AVCodecContext *avctx, - unsigned char *frame, int buf_size, void *data) -{ - 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 int MPA_encode_close(AVCodecContext *avctx) -{ - av_freep(&avctx->coded_frame); - return 0; -} - -AVCodec mp2_encoder = { - "mp2", - CODEC_TYPE_AUDIO, - CODEC_ID_MP2, - sizeof(MpegAudioContext), - MPA_encode_init, - MPA_encode_frame, - MPA_encode_close, - NULL, -}; - -#undef FIX
--- a/mpegaudio.h Sun May 20 15:11:55 2007 +0000 +++ b/mpegaudio.h Sun May 20 22:50:29 2007 +0000 @@ -26,6 +26,7 @@ #ifndef MPEGAUDIO_H #define MPEGAUDIO_H +#include "avcodec.h" #include "bitstream.h" #include "dsputil.h" @@ -115,7 +116,7 @@ AVCodecContext* avctx; } MPADecodeContext; -int l2_select_table(int bitrate, int nb_channels, int freq, int lsf); +int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf); int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate); void ff_mpa_synth_init(MPA_INT *window); void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
--- a/mpegaudiodec.c Sun May 20 15:11:55 2007 +0000 +++ b/mpegaudiodec.c Sun May 20 22:50:29 2007 +0000 @@ -1140,28 +1140,6 @@ return 12; } -/* bitrate is in kb/s */ -int l2_select_table(int bitrate, int nb_channels, int freq, int lsf) -{ - int ch_bitrate, table; - - ch_bitrate = bitrate / nb_channels; - if (!lsf) { - if ((freq == 48000 && ch_bitrate >= 56) || - (ch_bitrate >= 56 && ch_bitrate <= 80)) - table = 0; - else if (freq != 48000 && ch_bitrate >= 96) - table = 1; - else if (freq != 32000 && ch_bitrate <= 48) - table = 2; - else - table = 3; - } else { - table = 4; - } - return table; -} - static int mp_decode_layer2(MPADecodeContext *s) { int sblimit; /* number of used subbands */ @@ -1173,7 +1151,7 @@ int scale, qindex, bits, steps, k, l, m, b; /* select decoding table */ - table = l2_select_table(s->bit_rate / 1000, s->nb_channels, + table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels, s->sample_rate, s->lsf); sblimit = ff_mpa_sblimit_table[table]; alloc_table = ff_mpa_alloc_tables[table];
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/mpegaudioenc.c Sun May 20 22:50:29 2007 +0000 @@ -0,0 +1,802 @@ +/* + * The simplest mpeg audio layer 2 encoder + * Copyright (c) 2000, 2001 Fabrice Bellard. + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file mpegaudio.c + * The simplest mpeg audio layer 2 encoder. + */ + +#include "avcodec.h" +#include "bitstream.h" +#include "mpegaudio.h" + +/* currently, cannot change these constants (need to modify + quantization stage) */ +#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) +#define FIX(a) ((int)((a) * (1 << FRAC_BITS))) + +#define SAMPLES_BUF_SIZE 4096 + +typedef struct MpegAudioContext { + PutBitContext pb; + int nb_channels; + int freq, bit_rate; + int lsf; /* 1 if mpeg2 low bitrate selected */ + int bitrate_index; /* bit rate */ + int freq_index; + int frame_size; /* frame size, in bits, without padding */ + int64_t nb_samples; /* total number of samples encoded */ + /* padding computation */ + int frame_frac, frame_frac_incr, do_padding; + short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ + int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ + int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; + unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ + /* code to group 3 scale factors */ + unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; + int sblimit; /* number of used subbands */ + const unsigned char *alloc_table; +} MpegAudioContext; + +/* define it to use floats in quantization (I don't like floats !) */ +//#define USE_FLOATS + +#include "mpegaudiodata.h" +#include "mpegaudiotab.h" + +static int MPA_encode_init(AVCodecContext *avctx) +{ + 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; +} + +/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ +static void idct32(int *out, int *tab) +{ + 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]]; + } +} + +#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) + +static void filter(MpegAudioContext *s, int ch, short *samples, int incr) +{ + 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 compute_scale_factors(unsigned char scale_code[SBLIMIT], + unsigned char scale_factors[SBLIMIT][3], + int sb_samples[3][12][SBLIMIT], + int sblimit) +{ + 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 > 0) { + 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); //cant 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; + } +} + +/* The most important function : psycho acoustic module. In this + encoder there is basically none, so this is the worst you can do, + but also this is the simpler. */ +static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) +{ + int i; + + for(i=0;i<s->sblimit;i++) { + smr[i] = (int)(fixed_smr[i] * 10); + } +} + + +#define SB_NOTALLOCATED 0 +#define SB_ALLOCATED 1 +#define SB_NOMORE 2 + +/* Try to maximize the smr while using a number of bits inferior to + the frame size. I tried to make the code simpler, faster and + smaller than other encoders :-) */ +static void compute_bit_allocation(MpegAudioContext *s, + short smr1[MPA_MAX_CHANNELS][SBLIMIT], + unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], + int *padding) +{ + 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 = 0x80000000; + 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 +} + +/* + * Output the mpeg audio layer 2 frame. Note how the code is small + * compared to other encoders :-) + */ +static void encode_frame(MpegAudioContext *s, + unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], + int padding) +{ + 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 int MPA_encode_frame(AVCodecContext *avctx, + unsigned char *frame, int buf_size, void *data) +{ + 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 int MPA_encode_close(AVCodecContext *avctx) +{ + av_freep(&avctx->coded_frame); + return 0; +} + +AVCodec mp2_encoder = { + "mp2", + CODEC_TYPE_AUDIO, + CODEC_ID_MP2, + sizeof(MpegAudioContext), + MPA_encode_init, + MPA_encode_frame, + MPA_encode_close, + NULL, +}; + +#undef FIX