/* * The simplest mpeg audio layer 2 encoder * Copyright (c) 2000, 2001 Fabrice Bellard * * This file is part of Libav. * * Libav 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. * * Libav 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 Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * The simplest mpeg audio layer 2 encoder. */ #include "libavutil/channel_layout.h" #include "avcodec.h" #include "internal.h" #include "put_bits.h" #define FRAC_BITS 15 /* fractional bits for sb_samples and dct */ #define WFRAC_BITS 14 /* fractional bits for window */ #include "mpegaudio.h" #include "mpegaudiodsp.h" /* currently, cannot change these constants (need to modify quantization stage) */ #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) #define SAMPLES_BUF_SIZE 4096 typedef struct MpegAudioContext { PutBitContext pb; int nb_channels; 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 */ /* 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 av_cold 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 AVERROR(EINVAL); } bitrate = bitrate / 1000; s->nb_channels = channels; avctx->frame_size = MPA_FRAME_SIZE; avctx->delay = 512 - 32 + 1; /* encoding freq */ s->lsf = 0; for(i=0;i<3;i++) { if (avpriv_mpa_freq_tab[i] == freq) break; if ((avpriv_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 AVERROR(EINVAL); } s->freq_index = i; /* encoding bitrate & frequency */ for(i=0;i<15;i++) { if (avpriv_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 AVERROR(EINVAL); } 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]; av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", bitrate, freq, s->frame_size, table, s->frame_frac_incr); 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; } #if FF_API_OLD_ENCODE_AUDIO avctx->coded_frame= avcodec_alloc_frame(); if (!avctx->coded_frame) return AVERROR(ENOMEM); #endif 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, const short *samples, int incr) { short *p, *q; int sum, offset, i, j; int tmp[64]; int tmp1[32]; int *out; 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; } 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 > 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 */ } av_dlog(NULL, "%2d:%d in=%x %x %d\n", j, i, vmax, scale_factor_table[index], index); /* 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 */ } av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j, sf[0], sf[1], sf[2], d1, d2, code); 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 = 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 (max_sb < 0) break; av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n", current_frame_size, max_frame_size, max_sb, max_ch, bit_alloc[max_ch][max_sb]); /* 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); } /* * 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])); } else { 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, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { MpegAudioContext *s = avctx->priv_data; const int16_t *samples = (const int16_t *)frame->data[0]; short smr[MPA_MAX_CHANNELS][SBLIMIT]; unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; int padding, i, ret; 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); if ((ret = ff_alloc_packet(avpkt, MPA_MAX_CODED_FRAME_SIZE))) { av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); return ret; } init_put_bits(&s->pb, avpkt->data, avpkt->size); encode_frame(s, bit_alloc, padding); if (frame->pts != AV_NOPTS_VALUE) avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay); avpkt->size = put_bits_count(&s->pb) / 8; *got_packet_ptr = 1; return 0; } static av_cold int MPA_encode_close(AVCodecContext *avctx) { #if FF_API_OLD_ENCODE_AUDIO av_freep(&avctx->coded_frame); #endif return 0; } static const AVCodecDefault mp2_defaults[] = { { "b", "128k" }, { NULL }, }; AVCodec ff_mp2_encoder = { .name = "mp2", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP2, .priv_data_size = sizeof(MpegAudioContext), .init = MPA_encode_init, .encode2 = MPA_encode_frame, .close = MPA_encode_close, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .supported_samplerates = (const int[]){ 44100, 48000, 32000, 22050, 24000, 16000, 0 }, .channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, 0 }, .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), .defaults = mp2_defaults, };