786 lines
23 KiB
C
Executable File
786 lines
23 KiB
C
Executable File
/*
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* The simplest mpeg audio layer 2 encoder
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* Copyright (c) 2000, 2001 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* The simplest mpeg audio layer 2 encoder.
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*/
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#include "libavutil/channel_layout.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "put_bits.h"
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#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
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#define WFRAC_BITS 14 /* fractional bits for window */
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#include "mpegaudio.h"
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#include "mpegaudiodsp.h"
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#include "mpegaudiodata.h"
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#include "mpegaudiotab.h"
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/* currently, cannot change these constants (need to modify
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quantization stage) */
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#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
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#define SAMPLES_BUF_SIZE 4096
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typedef struct MpegAudioContext {
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PutBitContext pb;
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int nb_channels;
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int lsf; /* 1 if mpeg2 low bitrate selected */
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int bitrate_index; /* bit rate */
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int freq_index;
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int frame_size; /* frame size, in bits, without padding */
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/* padding computation */
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int frame_frac, frame_frac_incr, do_padding;
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short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
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int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
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int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
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unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
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/* code to group 3 scale factors */
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unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
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int sblimit; /* number of used subbands */
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const unsigned char *alloc_table;
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int16_t filter_bank[512];
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int scale_factor_table[64];
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unsigned char scale_diff_table[128];
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#if USE_FLOATS
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float scale_factor_inv_table[64];
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#else
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int8_t scale_factor_shift[64];
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unsigned short scale_factor_mult[64];
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#endif
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unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
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} MpegAudioContext;
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static av_cold int MPA_encode_init(AVCodecContext *avctx)
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{
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MpegAudioContext *s = avctx->priv_data;
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int freq = avctx->sample_rate;
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int bitrate = avctx->bit_rate;
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int channels = avctx->channels;
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int i, v, table;
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float a;
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if (channels <= 0 || channels > 2){
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av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
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return AVERROR(EINVAL);
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}
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bitrate = bitrate / 1000;
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s->nb_channels = channels;
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avctx->frame_size = MPA_FRAME_SIZE;
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avctx->initial_padding = 512 - 32 + 1;
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/* encoding freq */
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s->lsf = 0;
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for(i=0;i<3;i++) {
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if (avpriv_mpa_freq_tab[i] == freq)
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break;
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if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
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s->lsf = 1;
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break;
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}
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}
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if (i == 3){
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av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
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return AVERROR(EINVAL);
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}
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s->freq_index = i;
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/* encoding bitrate & frequency */
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for(i=1;i<15;i++) {
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if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
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break;
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}
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if (i == 15 && !avctx->bit_rate) {
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i = 14;
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bitrate = avpriv_mpa_bitrate_tab[s->lsf][1][i];
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avctx->bit_rate = bitrate * 1000;
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}
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if (i == 15){
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av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
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return AVERROR(EINVAL);
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}
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s->bitrate_index = i;
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/* compute total header size & pad bit */
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a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
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s->frame_size = ((int)a) * 8;
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/* frame fractional size to compute padding */
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s->frame_frac = 0;
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s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
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/* select the right allocation table */
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table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
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/* number of used subbands */
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s->sblimit = ff_mpa_sblimit_table[table];
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s->alloc_table = ff_mpa_alloc_tables[table];
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ff_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
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bitrate, freq, s->frame_size, table, s->frame_frac_incr);
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for(i=0;i<s->nb_channels;i++)
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s->samples_offset[i] = 0;
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for(i=0;i<257;i++) {
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int v;
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v = ff_mpa_enwindow[i];
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#if WFRAC_BITS != 16
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v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
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#endif
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s->filter_bank[i] = v;
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if ((i & 63) != 0)
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v = -v;
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if (i != 0)
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s->filter_bank[512 - i] = v;
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}
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for(i=0;i<64;i++) {
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v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
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if (v <= 0)
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v = 1;
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s->scale_factor_table[i] = v;
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#if USE_FLOATS
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s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
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#else
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#define P 15
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s->scale_factor_shift[i] = 21 - P - (i / 3);
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s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
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#endif
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}
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for(i=0;i<128;i++) {
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v = i - 64;
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if (v <= -3)
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v = 0;
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else if (v < 0)
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v = 1;
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else if (v == 0)
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v = 2;
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else if (v < 3)
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v = 3;
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else
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v = 4;
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s->scale_diff_table[i] = v;
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}
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for(i=0;i<17;i++) {
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v = ff_mpa_quant_bits[i];
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if (v < 0)
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v = -v;
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else
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v = v * 3;
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s->total_quant_bits[i] = 12 * v;
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}
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return 0;
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}
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/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
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static void idct32(int *out, int *tab)
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{
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int i, j;
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int *t, *t1, xr;
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const int *xp = costab32;
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for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
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t = tab + 30;
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t1 = tab + 2;
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do {
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t[0] += t[-4];
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t[1] += t[1 - 4];
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t -= 4;
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} while (t != t1);
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t = tab + 28;
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t1 = tab + 4;
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do {
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t[0] += t[-8];
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t[1] += t[1-8];
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t[2] += t[2-8];
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t[3] += t[3-8];
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t -= 8;
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} while (t != t1);
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t = tab;
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t1 = tab + 32;
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do {
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t[ 3] = -t[ 3];
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t[ 6] = -t[ 6];
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t[11] = -t[11];
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t[12] = -t[12];
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t[13] = -t[13];
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t[15] = -t[15];
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t += 16;
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} while (t != t1);
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t = tab;
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t1 = tab + 8;
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do {
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int x1, x2, x3, x4;
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x3 = MUL(t[16], FIX(M_SQRT2*0.5));
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x4 = t[0] - x3;
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x3 = t[0] + x3;
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x2 = MUL(-(t[24] + t[8]), FIX(M_SQRT2*0.5));
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x1 = MUL((t[8] - x2), xp[0]);
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x2 = MUL((t[8] + x2), xp[1]);
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t[ 0] = x3 + x1;
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t[ 8] = x4 - x2;
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t[16] = x4 + x2;
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t[24] = x3 - x1;
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t++;
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} while (t != t1);
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xp += 2;
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t = tab;
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t1 = tab + 4;
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do {
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xr = MUL(t[28],xp[0]);
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t[28] = (t[0] - xr);
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t[0] = (t[0] + xr);
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xr = MUL(t[4],xp[1]);
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t[ 4] = (t[24] - xr);
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t[24] = (t[24] + xr);
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xr = MUL(t[20],xp[2]);
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t[20] = (t[8] - xr);
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t[ 8] = (t[8] + xr);
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xr = MUL(t[12],xp[3]);
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t[12] = (t[16] - xr);
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t[16] = (t[16] + xr);
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t++;
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} while (t != t1);
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xp += 4;
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for (i = 0; i < 4; i++) {
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xr = MUL(tab[30-i*4],xp[0]);
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tab[30-i*4] = (tab[i*4] - xr);
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tab[ i*4] = (tab[i*4] + xr);
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xr = MUL(tab[ 2+i*4],xp[1]);
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tab[ 2+i*4] = (tab[28-i*4] - xr);
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tab[28-i*4] = (tab[28-i*4] + xr);
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xr = MUL(tab[31-i*4],xp[0]);
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tab[31-i*4] = (tab[1+i*4] - xr);
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tab[ 1+i*4] = (tab[1+i*4] + xr);
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xr = MUL(tab[ 3+i*4],xp[1]);
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tab[ 3+i*4] = (tab[29-i*4] - xr);
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tab[29-i*4] = (tab[29-i*4] + xr);
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xp += 2;
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}
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t = tab + 30;
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t1 = tab + 1;
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do {
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xr = MUL(t1[0], *xp);
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t1[0] = (t[0] - xr);
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t[0] = (t[0] + xr);
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t -= 2;
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t1 += 2;
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xp++;
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} while (t >= tab);
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for(i=0;i<32;i++) {
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out[i] = tab[bitinv32[i]];
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}
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}
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#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
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static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
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{
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short *p, *q;
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int sum, offset, i, j;
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int tmp[64];
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int tmp1[32];
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int *out;
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offset = s->samples_offset[ch];
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out = &s->sb_samples[ch][0][0][0];
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for(j=0;j<36;j++) {
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/* 32 samples at once */
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for(i=0;i<32;i++) {
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s->samples_buf[ch][offset + (31 - i)] = samples[0];
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samples += incr;
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}
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/* filter */
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p = s->samples_buf[ch] + offset;
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q = s->filter_bank;
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/* maxsum = 23169 */
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for(i=0;i<64;i++) {
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sum = p[0*64] * q[0*64];
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sum += p[1*64] * q[1*64];
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sum += p[2*64] * q[2*64];
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sum += p[3*64] * q[3*64];
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sum += p[4*64] * q[4*64];
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sum += p[5*64] * q[5*64];
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sum += p[6*64] * q[6*64];
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sum += p[7*64] * q[7*64];
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tmp[i] = sum;
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p++;
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q++;
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}
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tmp1[0] = tmp[16] >> WSHIFT;
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for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
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for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
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idct32(out, tmp1);
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/* advance of 32 samples */
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offset -= 32;
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out += 32;
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/* handle the wrap around */
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if (offset < 0) {
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memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
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s->samples_buf[ch], (512 - 32) * 2);
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offset = SAMPLES_BUF_SIZE - 512;
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}
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}
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s->samples_offset[ch] = offset;
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}
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static void compute_scale_factors(MpegAudioContext *s,
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unsigned char scale_code[SBLIMIT],
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unsigned char scale_factors[SBLIMIT][3],
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int sb_samples[3][12][SBLIMIT],
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int sblimit)
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{
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int *p, vmax, v, n, i, j, k, code;
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int index, d1, d2;
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unsigned char *sf = &scale_factors[0][0];
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for(j=0;j<sblimit;j++) {
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for(i=0;i<3;i++) {
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/* find the max absolute value */
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p = &sb_samples[i][0][j];
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vmax = abs(*p);
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for(k=1;k<12;k++) {
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p += SBLIMIT;
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v = abs(*p);
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if (v > vmax)
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vmax = v;
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}
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/* compute the scale factor index using log 2 computations */
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if (vmax > 1) {
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n = av_log2(vmax);
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/* n is the position of the MSB of vmax. now
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use at most 2 compares to find the index */
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index = (21 - n) * 3 - 3;
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if (index >= 0) {
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while (vmax <= s->scale_factor_table[index+1])
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index++;
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} else {
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index = 0; /* very unlikely case of overflow */
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}
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} else {
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index = 62; /* value 63 is not allowed */
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}
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ff_dlog(NULL, "%2d:%d in=%x %x %d\n",
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j, i, vmax, s->scale_factor_table[index], index);
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/* store the scale factor */
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av_assert2(index >=0 && index <= 63);
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sf[i] = index;
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}
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/* compute the transmission factor : look if the scale factors
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are close enough to each other */
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d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
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d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
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/* handle the 25 cases */
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switch(d1 * 5 + d2) {
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case 0*5+0:
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case 0*5+4:
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case 3*5+4:
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case 4*5+0:
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case 4*5+4:
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code = 0;
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break;
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case 0*5+1:
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case 0*5+2:
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case 4*5+1:
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case 4*5+2:
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code = 3;
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sf[2] = sf[1];
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break;
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case 0*5+3:
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case 4*5+3:
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code = 3;
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sf[1] = sf[2];
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break;
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case 1*5+0:
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case 1*5+4:
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case 2*5+4:
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code = 1;
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sf[1] = sf[0];
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break;
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case 1*5+1:
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case 1*5+2:
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case 2*5+0:
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case 2*5+1:
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case 2*5+2:
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code = 2;
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sf[1] = sf[2] = sf[0];
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break;
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case 2*5+3:
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case 3*5+3:
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code = 2;
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sf[0] = sf[1] = sf[2];
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break;
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case 3*5+0:
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case 3*5+1:
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case 3*5+2:
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code = 2;
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sf[0] = sf[2] = sf[1];
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break;
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case 1*5+3:
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code = 2;
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if (sf[0] > sf[2])
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sf[0] = sf[2];
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sf[1] = sf[2] = sf[0];
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break;
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default:
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av_assert2(0); //cannot happen
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code = 0; /* kill warning */
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}
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|
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ff_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
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sf[0], sf[1], sf[2], d1, d2, code);
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scale_code[j] = code;
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sf += 3;
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}
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}
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|
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/* 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;
|
|
ff_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 += s->total_quant_bits[alloc[1]];
|
|
} else {
|
|
/* increments bit allocation */
|
|
b = bit_alloc[max_ch][max_sb];
|
|
incr = s->total_quant_bits[alloc[b + 1]] -
|
|
s->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;
|
|
av_assert0(*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 = MPEG-1 ID, 0 = MPEG-2 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 */
|
|
#if USE_FLOATS
|
|
{
|
|
float a;
|
|
a = (float)sample * s->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 = s->scale_factor_shift[e];
|
|
mult = s->scale_factor_mult[e];
|
|
|
|
/* normalize to P bits */
|
|
if (shift < 0)
|
|
q1 = sample * (1 << -shift);
|
|
else
|
|
q1 = sample >> shift;
|
|
q1 = (q1 * mult) >> P;
|
|
q1 += 1 << P;
|
|
if (q1 < 0)
|
|
q1 = 0;
|
|
q[m] = (q1 * (unsigned)steps) >> (P + 1);
|
|
}
|
|
#endif
|
|
if (q[m] >= steps)
|
|
q[m] = steps - 1;
|
|
av_assert2(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, 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_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE, 0)) < 0)
|
|
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->initial_padding);
|
|
|
|
avpkt->size = put_bits_count(&s->pb) / 8;
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
static const AVCodecDefault mp2_defaults[] = {
|
|
{ "b", "0" },
|
|
{ NULL },
|
|
};
|
|
|