mirror of https://github.com/FFmpeg/FFmpeg.git
Originally committed as revision 9082 to svn://svn.ffmpeg.org/ffmpeg/trunkpull/126/head
Aurelien Jacobs
18 years ago
parent
ca6e50afc1
commit
08aa2c9bd2
5 changed files with 825 additions and 796 deletions
@ -0,0 +1,802 @@ |
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/*
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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 mpegaudio.c |
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* The simplest mpeg audio layer 2 encoder. |
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*/ |
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#include "avcodec.h" |
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#include "bitstream.h" |
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#include "mpegaudio.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 FIX(a) ((int)((a) * (1 << 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 freq, bit_rate; |
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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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int64_t nb_samples; /* total number of samples encoded */ |
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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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} MpegAudioContext; |
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/* define it to use floats in quantization (I don't like floats !) */ |
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//#define USE_FLOATS
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#include "mpegaudiodata.h" |
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#include "mpegaudiotab.h" |
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static 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 -1; |
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} |
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bitrate = bitrate / 1000; |
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s->nb_channels = channels; |
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s->freq = freq; |
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s->bit_rate = bitrate * 1000; |
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avctx->frame_size = MPA_FRAME_SIZE; |
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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 (ff_mpa_freq_tab[i] == freq) |
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break; |
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if ((ff_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 -1; |
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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=0;i<15;i++) { |
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if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate) |
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break; |
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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 -1; |
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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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#ifdef DEBUG |
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av_log(avctx, AV_LOG_DEBUG, "%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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#endif |
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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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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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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)(pow(2.0, (3 - i) / 3.0) * (1 << 20)); |
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if (v <= 0) |
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v = 1; |
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scale_factor_table[i] = v; |
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#ifdef USE_FLOATS |
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scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); |
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#else |
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#define P 15 |
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scale_factor_shift[i] = 21 - P - (i / 3); |
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scale_factor_mult[i] = (1 << P) * pow(2.0, (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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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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total_quant_bits[i] = 12 * v; |
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} |
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avctx->coded_frame= avcodec_alloc_frame(); |
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avctx->coded_frame->key_frame= 1; |
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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(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(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, 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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// print_pow1(samples, 1152);
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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 = 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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// print_pow(s->sb_samples, 1152);
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} |
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static void compute_scale_factors(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 > 0) { |
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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 <= 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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#if 0 |
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printf("%2d:%d in=%x %x %d\n", |
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j, i, vmax, scale_factor_table[index], index); |
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#endif |
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/* store the scale factor */ |
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assert(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 = scale_diff_table[sf[0] - sf[1] + 64]; |
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d2 = 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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assert(0); //cant happen
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code = 0; /* kill warning */ |
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} |
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#if 0 |
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printf("%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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#endif |
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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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/* The most important function : psycho acoustic module. In this
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encoder there is basically none, so this is the worst you can do, |
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but also this is the simpler. */ |
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static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) |
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{ |
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int i; |
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for(i=0;i<s->sblimit;i++) { |
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smr[i] = (int)(fixed_smr[i] * 10); |
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} |
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} |
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#define SB_NOTALLOCATED 0 |
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#define SB_ALLOCATED 1 |
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#define SB_NOMORE 2 |
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/* Try to maximize the smr while using a number of bits inferior to
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the frame size. I tried to make the code simpler, faster and |
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smaller than other encoders :-) */ |
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static void compute_bit_allocation(MpegAudioContext *s, |
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short smr1[MPA_MAX_CHANNELS][SBLIMIT], |
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unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], |
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int *padding) |
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{ |
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int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; |
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int incr; |
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short smr[MPA_MAX_CHANNELS][SBLIMIT]; |
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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 |
||||
Loading…
Reference in new issue