[FFmpeg-devel] [PATCH] avcodec: split mp2 encoder into float and fixed
Michael Niedermayer
michaelni at gmx.at
Sat Nov 16 13:35:43 CET 2013
This makes the USE_FLOATS == 0 available to the end user
More float optimizations can easily be added as well now
common code should be factored out into a common file once all
fixed point & floating point optimizations are done, this is to
avoid having to move code back and forth between files.
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
---
libavcodec/Makefile | 4 +-
libavcodec/allcodecs.c | 1 +
libavcodec/mpegaudioenc.c | 798 ------------------------------------
libavcodec/mpegaudioenc_fixed.c | 41 ++
libavcodec/mpegaudioenc_float.c | 42 ++
libavcodec/mpegaudioenc_template.c | 777 +++++++++++++++++++++++++++++++++++
tests/fate/acodec.mak | 4 +
tests/ref/acodec/mp2fixed | 4 +
8 files changed, 872 insertions(+), 799 deletions(-)
delete mode 100644 libavcodec/mpegaudioenc.c
create mode 100644 libavcodec/mpegaudioenc_fixed.c
create mode 100644 libavcodec/mpegaudioenc_float.c
create mode 100644 libavcodec/mpegaudioenc_template.c
create mode 100644 tests/ref/acodec/mp2fixed
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index d3b25bf..867f939 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -284,7 +284,9 @@ OBJS-$(CONFIG_MOVTEXT_ENCODER) += movtextenc.o ass_split.o
OBJS-$(CONFIG_MP1_DECODER) += mpegaudiodec.o
OBJS-$(CONFIG_MP1FLOAT_DECODER) += mpegaudiodec_float.o
OBJS-$(CONFIG_MP2_DECODER) += mpegaudiodec.o
-OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o \
+OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc_float.o mpegaudio.o \
+ mpegaudiodata.o mpegaudiodsp_data.o
+OBJS-$(CONFIG_MP2FIXED_ENCODER) += mpegaudioenc_fixed.o mpegaudio.o \
mpegaudiodata.o mpegaudiodsp_data.o
OBJS-$(CONFIG_MP2FLOAT_DECODER) += mpegaudiodec_float.o
OBJS-$(CONFIG_MP3_DECODER) += mpegaudiodec.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 2507eee..dab6d50 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -350,6 +350,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(MP1FLOAT, mp1float);
REGISTER_ENCDEC (MP2, mp2);
REGISTER_DECODER(MP2FLOAT, mp2float);
+ REGISTER_ENCODER(MP2FIXED, mp2fixed);
REGISTER_DECODER(MP3, mp3);
REGISTER_DECODER(MP3FLOAT, mp3float);
REGISTER_DECODER(MP3ADU, mp3adu);
diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
deleted file mode 100644
index 0054019..0000000
--- a/libavcodec/mpegaudioenc.c
+++ /dev/null
@@ -1,798 +0,0 @@
-/*
- * The simplest mpeg audio layer 2 encoder
- * Copyright (c) 2000, 2001 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * 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 */
-
-/* define it to use floats in quantization (I don't like floats !) */
-#define USE_FLOATS
-
-#include "mpegaudio.h"
-#include "mpegaudiodsp.h"
-#include "mpegaudiodata.h"
-#include "mpegaudiotab.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;
- int16_t filter_bank[512];
- int scale_factor_table[64];
- unsigned char scale_diff_table[128];
-#ifdef USE_FLOATS
- float scale_factor_inv_table[64];
-#else
- int8_t scale_factor_shift[64];
- unsigned short scale_factor_mult[64];
-#endif
- unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
-} MpegAudioContext;
-
-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
- s->filter_bank[i] = v;
- if ((i & 63) != 0)
- v = -v;
- if (i != 0)
- s->filter_bank[512 - i] = v;
- }
-
- for(i=0;i<64;i++) {
- v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
- if (v <= 0)
- v = 1;
- s->scale_factor_table[i] = v;
-#ifdef USE_FLOATS
- s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
-#else
-#define P 15
- s->scale_factor_shift[i] = 21 - P - (i / 3);
- s->scale_factor_mult[i] = (1 << P) * exp2((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;
- s->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;
- s->total_quant_bits[i] = 12 * v;
- }
-
- 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 = s->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(MpegAudioContext *s,
- 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 <= s->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, s->scale_factor_table[index], index);
- /* store the scale factor */
- av_assert2(index >=0 && index <= 63);
- sf[i] = index;
- }
-
- /* compute the transmission factor : look if the scale factors
- are close enough to each other */
- d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
- d2 = s->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:
- av_assert2(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 += 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 = 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 * 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 << (-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;
- 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)
- 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 const AVCodecDefault mp2_defaults[] = {
- { "b", "128k" },
- { NULL },
-};
-
-AVCodec ff_mp2_encoder = {
- .name = "mp2",
- .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_MP2,
- .priv_data_size = sizeof(MpegAudioContext),
- .init = MPA_encode_init,
- .encode2 = MPA_encode_frame,
- .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 },
- .defaults = mp2_defaults,
-};
diff --git a/libavcodec/mpegaudioenc_fixed.c b/libavcodec/mpegaudioenc_fixed.c
new file mode 100644
index 0000000..98c0bbe
--- /dev/null
+++ b/libavcodec/mpegaudioenc_fixed.c
@@ -0,0 +1,41 @@
+/*
+ * The simplest mpeg audio layer 2 encoder
+ * Copyright (c) 2000, 2001 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "mpegaudioenc_template.c"
+
+AVCodec ff_mp2fixed_encoder = {
+ .name = "mp2fixed",
+ .long_name = NULL_IF_CONFIG_SMALL("MP2 fixed point (MPEG audio layer 2)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_MP2,
+ .priv_data_size = sizeof(MpegAudioContext),
+ .init = MPA_encode_init,
+ .encode2 = MPA_encode_frame,
+ .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 },
+ .defaults = mp2_defaults,
+};
\ No newline at end of file
diff --git a/libavcodec/mpegaudioenc_float.c b/libavcodec/mpegaudioenc_float.c
new file mode 100644
index 0000000..7712307
--- /dev/null
+++ b/libavcodec/mpegaudioenc_float.c
@@ -0,0 +1,42 @@
+/*
+ * The simplest mpeg audio layer 2 encoder
+ * Copyright (c) 2000, 2001 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define USE_FLOATS
+#include "mpegaudioenc_template.c"
+
+AVCodec ff_mp2_encoder = {
+ .name = "mp2",
+ .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_MP2,
+ .priv_data_size = sizeof(MpegAudioContext),
+ .init = MPA_encode_init,
+ .encode2 = MPA_encode_frame,
+ .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 },
+ .defaults = mp2_defaults,
+};
diff --git a/libavcodec/mpegaudioenc_template.c b/libavcodec/mpegaudioenc_template.c
new file mode 100644
index 0000000..11067b5
--- /dev/null
+++ b/libavcodec/mpegaudioenc_template.c
@@ -0,0 +1,777 @@
+/*
+ * The simplest mpeg audio layer 2 encoder
+ * Copyright (c) 2000, 2001 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * 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"
+#include "mpegaudiodata.h"
+#include "mpegaudiotab.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;
+ int16_t filter_bank[512];
+ int scale_factor_table[64];
+ unsigned char scale_diff_table[128];
+#ifdef USE_FLOATS
+ float scale_factor_inv_table[64];
+#else
+ int8_t scale_factor_shift[64];
+ unsigned short scale_factor_mult[64];
+#endif
+ unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
+} MpegAudioContext;
+
+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
+ s->filter_bank[i] = v;
+ if ((i & 63) != 0)
+ v = -v;
+ if (i != 0)
+ s->filter_bank[512 - i] = v;
+ }
+
+ for(i=0;i<64;i++) {
+ v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
+ if (v <= 0)
+ v = 1;
+ s->scale_factor_table[i] = v;
+#ifdef USE_FLOATS
+ s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
+#else
+#define P 15
+ s->scale_factor_shift[i] = 21 - P - (i / 3);
+ s->scale_factor_mult[i] = (1 << P) * exp2((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;
+ s->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;
+ s->total_quant_bits[i] = 12 * v;
+ }
+
+ 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 = s->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(MpegAudioContext *s,
+ 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 <= s->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, s->scale_factor_table[index], index);
+ /* store the scale factor */
+ av_assert2(index >=0 && index <= 63);
+ sf[i] = index;
+ }
+
+ /* compute the transmission factor : look if the scale factors
+ are close enough to each other */
+ d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
+ d2 = s->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:
+ av_assert2(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 += 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 = 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 * 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 << (-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;
+ 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)
+ 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 const AVCodecDefault mp2_defaults[] = {
+ { "b", "128k" },
+ { NULL },
+};
+
diff --git a/tests/fate/acodec.mak b/tests/fate/acodec.mak
index 78508bc..10d4392 100644
--- a/tests/fate/acodec.mak
+++ b/tests/fate/acodec.mak
@@ -68,6 +68,10 @@ FATE_ACODEC-$(call ENCDEC, MP2, MP2 MP3) += fate-acodec-mp2
fate-acodec-mp2: FMT = mp2
fate-acodec-mp2: CMP_SHIFT = -1924
+FATE_ACODEC-$(call ENCDEC, MP2FIXED MP2 , MP2 MP3) += fate-acodec-mp2fixed
+fate-acodec-mp2fixed: FMT = mp2
+fate-acodec-mp2fixed: CMP_SHIFT = -1924
+
FATE_ACODEC-$(call ENCDEC, ALAC, MOV) += fate-acodec-alac
fate-acodec-alac: FMT = mov
fate-acodec-alac: CODEC = alac -compression_level 1
diff --git a/tests/ref/acodec/mp2fixed b/tests/ref/acodec/mp2fixed
new file mode 100644
index 0000000..0203014
--- /dev/null
+++ b/tests/ref/acodec/mp2fixed
@@ -0,0 +1,4 @@
+28fbc7485c7939f40368f79adccb3e3d *tests/data/fate/acodec-mp2fixed.mp2
+96130 tests/data/fate/acodec-mp2fixed.mp2
+87461bd4ce4b0e0cbbf6c43621baf261 *tests/data/fate/acodec-mp2fixed.out.wav
+stddev: 4384.26 PSNR: 23.49 MAXDIFF:52632 bytes: 1058400/ 1057916
--
1.7.9.5
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