remove dependency of mpeg audio encoder over mpeg audio decoder
Originally committed as revision 9082 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
parent
ca6e50afc1
commit
08aa2c9bd2
@ -19,7 +19,7 @@ OBJS= bitstream.o \
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motion_est.o \
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imgconvert.o \
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mpeg12.o \
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mpegaudiodec.o mpegaudiodecheader.o mpegaudiodata.o \
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mpegaudiodec.o mpegaudiodecheader.o mpegaudiodata.o mpegaudio.o \
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simple_idct.o \
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ratecontrol.o \
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eval.o \
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@ -108,7 +108,7 @@ OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o mjpeg.o
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OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o mpegvideo.o
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OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o mjpegdec.o mjpeg.o
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OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o
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OBJS-$(CONFIG_MP2_ENCODER) += mpegaudio.o mpegaudiodata.o
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OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o mpegaudiodata.o
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OBJS-$(CONFIG_MPC7_DECODER) += mpc.o
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OBJS-$(CONFIG_MSMPEG4V1_DECODER) += msmpeg4.o msmpeg4data.o
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OBJS-$(CONFIG_MSMPEG4V1_ENCODER) += msmpeg4.o msmpeg4data.o
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@ -1,6 +1,6 @@
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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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* MPEG Audio common code
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* Copyright (c) 2001, 2002 Fabrice Bellard.
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*
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* This file is part of FFmpeg.
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*
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@ -21,782 +21,30 @@
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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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* MPEG Audio common code.
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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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/* bitrate is in kb/s */
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int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf)
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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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int ch_bitrate, table;
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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 = 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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ch_bitrate = bitrate / nb_channels;
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if (!lsf) {
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if ((freq == 48000 && ch_bitrate >= 56) ||
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(ch_bitrate >= 56 && ch_bitrate <= 80))
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table = 0;
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else if (freq != 48000 && ch_bitrate >= 96)
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table = 1;
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else if (freq != 32000 && ch_bitrate <= 48)
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table = 2;
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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;
|
||||
case 0*5+1:
|
||||
case 0*5+2:
|
||||
case 4*5+1:
|
||||
case 4*5+2:
|
||||
code = 3;
|
||||
sf[2] = sf[1];
|
||||
break;
|
||||
case 0*5+3:
|
||||
case 4*5+3:
|
||||
code = 3;
|
||||
sf[1] = sf[2];
|
||||
break;
|
||||
case 1*5+0:
|
||||
case 1*5+4:
|
||||
case 2*5+4:
|
||||
code = 1;
|
||||
sf[1] = sf[0];
|
||||
break;
|
||||
case 1*5+1:
|
||||
case 1*5+2:
|
||||
case 2*5+0:
|
||||
case 2*5+1:
|
||||
case 2*5+2:
|
||||
code = 2;
|
||||
sf[1] = sf[2] = sf[0];
|
||||
break;
|
||||
case 2*5+3:
|
||||
case 3*5+3:
|
||||
code = 2;
|
||||
sf[0] = sf[1] = sf[2];
|
||||
break;
|
||||
case 3*5+0:
|
||||
case 3*5+1:
|
||||
case 3*5+2:
|
||||
code = 2;
|
||||
sf[0] = sf[2] = sf[1];
|
||||
break;
|
||||
case 1*5+3:
|
||||
code = 2;
|
||||
if (sf[0] > sf[2])
|
||||
sf[0] = sf[2];
|
||||
sf[1] = sf[2] = sf[0];
|
||||
break;
|
||||
default:
|
||||
assert(0); //cant happen
|
||||
code = 0; /* kill warning */
|
||||
}
|
||||
|
||||
#if 0
|
||||
printf("%d: %2d %2d %2d %d %d -> %d\n", j,
|
||||
sf[0], sf[1], sf[2], d1, d2, code);
|
||||
#endif
|
||||
scale_code[j] = code;
|
||||
sf += 3;
|
||||
}
|
||||
}
|
||||
|
||||
/* The most important function : psycho acoustic module. In this
|
||||
encoder there is basically none, so this is the worst you can do,
|
||||
but also this is the simpler. */
|
||||
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
|
||||
{
|
||||
int i;
|
||||
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
smr[i] = (int)(fixed_smr[i] * 10);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
#define SB_NOTALLOCATED 0
|
||||
#define SB_ALLOCATED 1
|
||||
#define SB_NOMORE 2
|
||||
|
||||
/* Try to maximize the smr while using a number of bits inferior to
|
||||
the frame size. I tried to make the code simpler, faster and
|
||||
smaller than other encoders :-) */
|
||||
static void compute_bit_allocation(MpegAudioContext *s,
|
||||
short smr1[MPA_MAX_CHANNELS][SBLIMIT],
|
||||
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
|
||||
int *padding)
|
||||
{
|
||||
int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
|
||||
int incr;
|
||||
short smr[MPA_MAX_CHANNELS][SBLIMIT];
|
||||
unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
|
||||
const unsigned char *alloc;
|
||||
|
||||
memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
|
||||
memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
|
||||
memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
|
||||
|
||||
/* compute frame size and padding */
|
||||
max_frame_size = s->frame_size;
|
||||
s->frame_frac += s->frame_frac_incr;
|
||||
if (s->frame_frac >= 65536) {
|
||||
s->frame_frac -= 65536;
|
||||
s->do_padding = 1;
|
||||
max_frame_size += 8;
|
||||
table = 3;
|
||||
} else {
|
||||
s->do_padding = 0;
|
||||
table = 4;
|
||||
}
|
||||
|
||||
/* 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
|
||||
return table;
|
||||
}
|
||||
|
||||
/*
|
||||
* 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
|
||||
|
@ -26,6 +26,7 @@
|
||||
#ifndef MPEGAUDIO_H
|
||||
#define MPEGAUDIO_H
|
||||
|
||||
#include "avcodec.h"
|
||||
#include "bitstream.h"
|
||||
#include "dsputil.h"
|
||||
|
||||
@ -115,7 +116,7 @@ typedef struct MPADecodeContext {
|
||||
AVCodecContext* avctx;
|
||||
} MPADecodeContext;
|
||||
|
||||
int l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
|
||||
int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
|
||||
int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate);
|
||||
void ff_mpa_synth_init(MPA_INT *window);
|
||||
void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
|
||||
|
@ -1140,28 +1140,6 @@ static int mp_decode_layer1(MPADecodeContext *s)
|
||||
return 12;
|
||||
}
|
||||
|
||||
/* bitrate is in kb/s */
|
||||
int l2_select_table(int bitrate, int nb_channels, int freq, int lsf)
|
||||
{
|
||||
int ch_bitrate, table;
|
||||
|
||||
ch_bitrate = bitrate / nb_channels;
|
||||
if (!lsf) {
|
||||
if ((freq == 48000 && ch_bitrate >= 56) ||
|
||||
(ch_bitrate >= 56 && ch_bitrate <= 80))
|
||||
table = 0;
|
||||
else if (freq != 48000 && ch_bitrate >= 96)
|
||||
table = 1;
|
||||
else if (freq != 32000 && ch_bitrate <= 48)
|
||||
table = 2;
|
||||
else
|
||||
table = 3;
|
||||
} else {
|
||||
table = 4;
|
||||
}
|
||||
return table;
|
||||
}
|
||||
|
||||
static int mp_decode_layer2(MPADecodeContext *s)
|
||||
{
|
||||
int sblimit; /* number of used subbands */
|
||||
@ -1173,7 +1151,7 @@ static int mp_decode_layer2(MPADecodeContext *s)
|
||||
int scale, qindex, bits, steps, k, l, m, b;
|
||||
|
||||
/* select decoding table */
|
||||
table = l2_select_table(s->bit_rate / 1000, s->nb_channels,
|
||||
table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
|
||||
s->sample_rate, s->lsf);
|
||||
sblimit = ff_mpa_sblimit_table[table];
|
||||
alloc_table = ff_mpa_alloc_tables[table];
|
||||
|
802
libavcodec/mpegaudioenc.c
Normal file
802
libavcodec/mpegaudioenc.c
Normal file
@ -0,0 +1,802 @@
|
||||
/*
|
||||
* The simplest mpeg audio layer 2 encoder
|
||||
* Copyright (c) 2000, 2001 Fabrice Bellard.
|
||||
*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* FFmpeg is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
/**
|
||||
* @file mpegaudio.c
|
||||
* The simplest mpeg audio layer 2 encoder.
|
||||
*/
|
||||
|
||||
#include "avcodec.h"
|
||||
#include "bitstream.h"
|
||||
#include "mpegaudio.h"
|
||||
|
||||
/* currently, cannot change these constants (need to modify
|
||||
quantization stage) */
|
||||
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
|
||||
#define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
|
||||
|
||||
#define SAMPLES_BUF_SIZE 4096
|
||||
|
||||
typedef struct MpegAudioContext {
|
||||
PutBitContext pb;
|
||||
int nb_channels;
|
||||
int freq, bit_rate;
|
||||
int lsf; /* 1 if mpeg2 low bitrate selected */
|
||||
int bitrate_index; /* bit rate */
|
||||
int freq_index;
|
||||
int frame_size; /* frame size, in bits, without padding */
|
||||
int64_t nb_samples; /* total number of samples encoded */
|
||||
/* padding computation */
|
||||
int frame_frac, frame_frac_incr, do_padding;
|
||||
short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
|
||||
int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
|
||||
int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
|
||||
unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
|
||||
/* code to group 3 scale factors */
|
||||
unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
|
||||
int sblimit; /* number of used subbands */
|
||||
const unsigned char *alloc_table;
|
||||
} MpegAudioContext;
|
||||
|
||||
/* define it to use floats in quantization (I don't like floats !) */
|
||||
//#define USE_FLOATS
|
||||
|
||||
#include "mpegaudiodata.h"
|
||||
#include "mpegaudiotab.h"
|
||||
|
||||
static int MPA_encode_init(AVCodecContext *avctx)
|
||||
{
|
||||
MpegAudioContext *s = avctx->priv_data;
|
||||
int freq = avctx->sample_rate;
|
||||
int bitrate = avctx->bit_rate;
|
||||
int channels = avctx->channels;
|
||||
int i, v, table;
|
||||
float a;
|
||||
|
||||
if (channels <= 0 || channels > 2){
|
||||
av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
|
||||
return -1;
|
||||
}
|
||||
bitrate = bitrate / 1000;
|
||||
s->nb_channels = channels;
|
||||
s->freq = freq;
|
||||
s->bit_rate = bitrate * 1000;
|
||||
avctx->frame_size = MPA_FRAME_SIZE;
|
||||
|
||||
/* encoding freq */
|
||||
s->lsf = 0;
|
||||
for(i=0;i<3;i++) {
|
||||
if (ff_mpa_freq_tab[i] == freq)
|
||||
break;
|
||||
if ((ff_mpa_freq_tab[i] / 2) == freq) {
|
||||
s->lsf = 1;
|
||||
break;
|
||||
}
|
||||
}
|
||||
if (i == 3){
|
||||
av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
|
||||
return -1;
|
||||
}
|
||||
s->freq_index = i;
|
||||
|
||||
/* encoding bitrate & frequency */
|
||||
for(i=0;i<15;i++) {
|
||||
if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
|
||||
break;
|
||||
}
|
||||
if (i == 15){
|
||||
av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
|
||||
return -1;
|
||||
}
|
||||
s->bitrate_index = i;
|
||||
|
||||
/* compute total header size & pad bit */
|
||||
|
||||
a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
|
||||
s->frame_size = ((int)a) * 8;
|
||||
|
||||
/* frame fractional size to compute padding */
|
||||
s->frame_frac = 0;
|
||||
s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
|
||||
|
||||
/* select the right allocation table */
|
||||
table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
|
||||
|
||||
/* number of used subbands */
|
||||
s->sblimit = ff_mpa_sblimit_table[table];
|
||||
s->alloc_table = ff_mpa_alloc_tables[table];
|
||||
|
||||
#ifdef DEBUG
|
||||
av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
|
||||
bitrate, freq, s->frame_size, table, s->frame_frac_incr);
|
||||
#endif
|
||||
|
||||
for(i=0;i<s->nb_channels;i++)
|
||||
s->samples_offset[i] = 0;
|
||||
|
||||
for(i=0;i<257;i++) {
|
||||
int v;
|
||||
v = ff_mpa_enwindow[i];
|
||||
#if WFRAC_BITS != 16
|
||||
v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
|
||||
#endif
|
||||
filter_bank[i] = v;
|
||||
if ((i & 63) != 0)
|
||||
v = -v;
|
||||
if (i != 0)
|
||||
filter_bank[512 - i] = v;
|
||||
}
|
||||
|
||||
for(i=0;i<64;i++) {
|
||||
v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
|
||||
if (v <= 0)
|
||||
v = 1;
|
||||
scale_factor_table[i] = v;
|
||||
#ifdef USE_FLOATS
|
||||
scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
|
||||
#else
|
||||
#define P 15
|
||||
scale_factor_shift[i] = 21 - P - (i / 3);
|
||||
scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
|
||||
#endif
|
||||
}
|
||||
for(i=0;i<128;i++) {
|
||||
v = i - 64;
|
||||
if (v <= -3)
|
||||
v = 0;
|
||||
else if (v < 0)
|
||||
v = 1;
|
||||
else if (v == 0)
|
||||
v = 2;
|
||||
else if (v < 3)
|
||||
v = 3;
|
||||
else
|
||||
v = 4;
|
||||
scale_diff_table[i] = v;
|
||||
}
|
||||
|
||||
for(i=0;i<17;i++) {
|
||||
v = ff_mpa_quant_bits[i];
|
||||
if (v < 0)
|
||||
v = -v;
|
||||
else
|
||||
v = v * 3;
|
||||
total_quant_bits[i] = 12 * v;
|
||||
}
|
||||
|
||||
avctx->coded_frame= avcodec_alloc_frame();
|
||||
avctx->coded_frame->key_frame= 1;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
|
||||
static void idct32(int *out, int *tab)
|
||||
{
|
||||
int i, j;
|
||||
int *t, *t1, xr;
|
||||
const int *xp = costab32;
|
||||
|
||||
for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
|
||||
|
||||
t = tab + 30;
|
||||
t1 = tab + 2;
|
||||
do {
|
||||
t[0] += t[-4];
|
||||
t[1] += t[1 - 4];
|
||||
t -= 4;
|
||||
} while (t != t1);
|
||||
|
||||
t = tab + 28;
|
||||
t1 = tab + 4;
|
||||
do {
|
||||
t[0] += t[-8];
|
||||
t[1] += t[1-8];
|
||||
t[2] += t[2-8];
|
||||
t[3] += t[3-8];
|
||||
t -= 8;
|
||||
} while (t != t1);
|
||||
|
||||
t = tab;
|
||||
t1 = tab + 32;
|
||||
do {
|
||||
t[ 3] = -t[ 3];
|
||||
t[ 6] = -t[ 6];
|
||||
|
||||
t[11] = -t[11];
|
||||
t[12] = -t[12];
|
||||
t[13] = -t[13];
|
||||
t[15] = -t[15];
|
||||
t += 16;
|
||||
} while (t != t1);
|
||||
|
||||
|
||||
t = tab;
|
||||
t1 = tab + 8;
|
||||
do {
|
||||
int x1, x2, x3, x4;
|
||||
|
||||
x3 = MUL(t[16], FIX(SQRT2*0.5));
|
||||
x4 = t[0] - x3;
|
||||
x3 = t[0] + x3;
|
||||
|
||||
x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
|
||||
x1 = MUL((t[8] - x2), xp[0]);
|
||||
x2 = MUL((t[8] + x2), xp[1]);
|
||||
|
||||
t[ 0] = x3 + x1;
|
||||
t[ 8] = x4 - x2;
|
||||
t[16] = x4 + x2;
|
||||
t[24] = x3 - x1;
|
||||
t++;
|
||||
} while (t != t1);
|
||||
|
||||
xp += 2;
|
||||
t = tab;
|
||||
t1 = tab + 4;
|
||||
do {
|
||||
xr = MUL(t[28],xp[0]);
|
||||
t[28] = (t[0] - xr);
|
||||
t[0] = (t[0] + xr);
|
||||
|
||||
xr = MUL(t[4],xp[1]);
|
||||
t[ 4] = (t[24] - xr);
|
||||
t[24] = (t[24] + xr);
|
||||
|
||||
xr = MUL(t[20],xp[2]);
|
||||
t[20] = (t[8] - xr);
|
||||
t[ 8] = (t[8] + xr);
|
||||
|
||||
xr = MUL(t[12],xp[3]);
|
||||
t[12] = (t[16] - xr);
|
||||
t[16] = (t[16] + xr);
|
||||
t++;
|
||||
} while (t != t1);
|
||||
xp += 4;
|
||||
|
||||
for (i = 0; i < 4; i++) {
|
||||
xr = MUL(tab[30-i*4],xp[0]);
|
||||
tab[30-i*4] = (tab[i*4] - xr);
|
||||
tab[ i*4] = (tab[i*4] + xr);
|
||||
|
||||
xr = MUL(tab[ 2+i*4],xp[1]);
|
||||
tab[ 2+i*4] = (tab[28-i*4] - xr);
|
||||
tab[28-i*4] = (tab[28-i*4] + xr);
|
||||
|
||||
xr = MUL(tab[31-i*4],xp[0]);
|
||||
tab[31-i*4] = (tab[1+i*4] - xr);
|
||||
tab[ 1+i*4] = (tab[1+i*4] + xr);
|
||||
|
||||
xr = MUL(tab[ 3+i*4],xp[1]);
|
||||
tab[ 3+i*4] = (tab[29-i*4] - xr);
|
||||
tab[29-i*4] = (tab[29-i*4] + xr);
|
||||
|
||||
xp += 2;
|
||||
}
|
||||
|
||||
t = tab + 30;
|
||||
t1 = tab + 1;
|
||||
do {
|
||||
xr = MUL(t1[0], *xp);
|
||||
t1[0] = (t[0] - xr);
|
||||
t[0] = (t[0] + xr);
|
||||
t -= 2;
|
||||
t1 += 2;
|
||||
xp++;
|
||||
} while (t >= tab);
|
||||
|
||||
for(i=0;i<32;i++) {
|
||||
out[i] = tab[bitinv32[i]];
|
||||
}
|
||||
}
|
||||
|
||||
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
|
||||
|
||||
static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
|
||||
{
|
||||
short *p, *q;
|
||||
int sum, offset, i, j;
|
||||
int tmp[64];
|
||||
int tmp1[32];
|
||||
int *out;
|
||||
|
||||
// print_pow1(samples, 1152);
|
||||
|
||||
offset = s->samples_offset[ch];
|
||||
out = &s->sb_samples[ch][0][0][0];
|
||||
for(j=0;j<36;j++) {
|
||||
/* 32 samples at once */
|
||||
for(i=0;i<32;i++) {
|
||||
s->samples_buf[ch][offset + (31 - i)] = samples[0];
|
||||
samples += incr;
|
||||
}
|
||||
|
||||
/* filter */
|
||||
p = s->samples_buf[ch] + offset;
|
||||
q = filter_bank;
|
||||
/* maxsum = 23169 */
|
||||
for(i=0;i<64;i++) {
|
||||
sum = p[0*64] * q[0*64];
|
||||
sum += p[1*64] * q[1*64];
|
||||
sum += p[2*64] * q[2*64];
|
||||
sum += p[3*64] * q[3*64];
|
||||
sum += p[4*64] * q[4*64];
|
||||
sum += p[5*64] * q[5*64];
|
||||
sum += p[6*64] * q[6*64];
|
||||
sum += p[7*64] * q[7*64];
|
||||
tmp[i] = sum;
|
||||
p++;
|
||||
q++;
|
||||
}
|
||||
tmp1[0] = tmp[16] >> WSHIFT;
|
||||
for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
|
||||
for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
|
||||
|
||||
idct32(out, tmp1);
|
||||
|
||||
/* advance of 32 samples */
|
||||
offset -= 32;
|
||||
out += 32;
|
||||
/* handle the wrap around */
|
||||
if (offset < 0) {
|
||||
memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
|
||||
s->samples_buf[ch], (512 - 32) * 2);
|
||||
offset = SAMPLES_BUF_SIZE - 512;
|
||||
}
|
||||
}
|
||||
s->samples_offset[ch] = offset;
|
||||
|
||||
// print_pow(s->sb_samples, 1152);
|
||||
}
|
||||
|
||||
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
|
||||
unsigned char scale_factors[SBLIMIT][3],
|
||||
int sb_samples[3][12][SBLIMIT],
|
||||
int sblimit)
|
||||
{
|
||||
int *p, vmax, v, n, i, j, k, code;
|
||||
int index, d1, d2;
|
||||
unsigned char *sf = &scale_factors[0][0];
|
||||
|
||||
for(j=0;j<sblimit;j++) {
|
||||
for(i=0;i<3;i++) {
|
||||
/* find the max absolute value */
|
||||
p = &sb_samples[i][0][j];
|
||||
vmax = abs(*p);
|
||||
for(k=1;k<12;k++) {
|
||||
p += SBLIMIT;
|
||||
v = abs(*p);
|
||||
if (v > vmax)
|
||||
vmax = v;
|
||||
}
|
||||
/* compute the scale factor index using log 2 computations */
|
||||
if (vmax > 0) {
|
||||
n = av_log2(vmax);
|
||||
/* n is the position of the MSB of vmax. now
|
||||
use at most 2 compares to find the index */
|
||||
index = (21 - n) * 3 - 3;
|
||||
if (index >= 0) {
|
||||
while (vmax <= scale_factor_table[index+1])
|
||||
index++;
|
||||
} else {
|
||||
index = 0; /* very unlikely case of overflow */
|
||||
}
|
||||
} else {
|
||||
index = 62; /* value 63 is not allowed */
|
||||
}
|
||||
|
||||
#if 0
|
||||
printf("%2d:%d in=%x %x %d\n",
|
||||
j, i, vmax, scale_factor_table[index], index);
|
||||
#endif
|
||||
/* store the scale factor */
|
||||
assert(index >=0 && index <= 63);
|
||||
sf[i] = index;
|
||||
}
|
||||
|
||||
/* compute the transmission factor : look if the scale factors
|
||||
are close enough to each other */
|
||||
d1 = scale_diff_table[sf[0] - sf[1] + 64];
|
||||
d2 = scale_diff_table[sf[1] - sf[2] + 64];
|
||||
|
||||
/* handle the 25 cases */
|
||||
switch(d1 * 5 + d2) {
|
||||
case 0*5+0:
|
||||
case 0*5+4:
|
||||
case 3*5+4:
|
||||
case 4*5+0:
|
||||
case 4*5+4:
|
||||
code = 0;
|
||||
break;
|
||||
case 0*5+1:
|
||||
case 0*5+2:
|
||||
case 4*5+1:
|
||||
case 4*5+2:
|
||||
code = 3;
|
||||
sf[2] = sf[1];
|
||||
break;
|
||||
case 0*5+3:
|
||||
case 4*5+3:
|
||||
code = 3;
|
||||
sf[1] = sf[2];
|
||||
break;
|
||||
case 1*5+0:
|
||||
case 1*5+4:
|
||||
case 2*5+4:
|
||||
code = 1;
|
||||
sf[1] = sf[0];
|
||||
break;
|
||||
case 1*5+1:
|
||||
case 1*5+2:
|
||||
case 2*5+0:
|
||||
case 2*5+1:
|
||||
case 2*5+2:
|
||||
code = 2;
|
||||
sf[1] = sf[2] = sf[0];
|
||||
break;
|
||||
case 2*5+3:
|
||||
case 3*5+3:
|
||||
code = 2;
|
||||
sf[0] = sf[1] = sf[2];
|
||||
break;
|
||||
case 3*5+0:
|
||||
case 3*5+1:
|
||||
case 3*5+2:
|
||||
code = 2;
|
||||
sf[0] = sf[2] = sf[1];
|
||||
break;
|
||||
case 1*5+3:
|
||||
code = 2;
|
||||
if (sf[0] > sf[2])
|
||||
sf[0] = sf[2];
|
||||
sf[1] = sf[2] = sf[0];
|
||||
break;
|
||||
default:
|
||||
assert(0); //cant happen
|
||||
code = 0; /* kill warning */
|
||||
}
|
||||
|
||||
#if 0
|
||||
printf("%d: %2d %2d %2d %d %d -> %d\n", j,
|
||||
sf[0], sf[1], sf[2], d1, d2, code);
|
||||
#endif
|
||||
scale_code[j] = code;
|
||||
sf += 3;
|
||||
}
|
||||
}
|
||||
|
||||
/* The most important function : psycho acoustic module. In this
|
||||
encoder there is basically none, so this is the worst you can do,
|
||||
but also this is the simpler. */
|
||||
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
|
||||
{
|
||||
int i;
|
||||
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
smr[i] = (int)(fixed_smr[i] * 10);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
#define SB_NOTALLOCATED 0
|
||||
#define SB_ALLOCATED 1
|
||||
#define SB_NOMORE 2
|
||||
|
||||
/* Try to maximize the smr while using a number of bits inferior to
|
||||
the frame size. I tried to make the code simpler, faster and
|
||||
smaller than other encoders :-) */
|
||||
static void compute_bit_allocation(MpegAudioContext *s,
|
||||
short smr1[MPA_MAX_CHANNELS][SBLIMIT],
|
||||
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
|
||||
int *padding)
|
||||
{
|
||||
int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
|
||||
int incr;
|
||||
short smr[MPA_MAX_CHANNELS][SBLIMIT];
|
||||
unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
|
||||
const unsigned char *alloc;
|
||||
|
||||
memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
|
||||
memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
|
||||
memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
|
||||
|
||||
/* compute frame size and padding */
|
||||
max_frame_size = s->frame_size;
|
||||
s->frame_frac += s->frame_frac_incr;
|
||||
if (s->frame_frac >= 65536) {
|
||||
s->frame_frac -= 65536;
|
||||
s->do_padding = 1;
|
||||
max_frame_size += 8;
|
||||
} else {
|
||||
s->do_padding = 0;
|
||||
}
|
||||
|
||||
/* compute the header + bit alloc size */
|
||||
current_frame_size = 32;
|
||||
alloc = s->alloc_table;
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
incr = alloc[0];
|
||||
current_frame_size += incr * s->nb_channels;
|
||||
alloc += 1 << incr;
|
||||
}
|
||||
for(;;) {
|
||||
/* look for the subband with the largest signal to mask ratio */
|
||||
max_sb = -1;
|
||||
max_ch = -1;
|
||||
max_smr = 0x80000000;
|
||||
for(ch=0;ch<s->nb_channels;ch++) {
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
|
||||
max_smr = smr[ch][i];
|
||||
max_sb = i;
|
||||
max_ch = ch;
|
||||
}
|
||||
}
|
||||
}
|
||||
#if 0
|
||||
printf("current=%d max=%d max_sb=%d alloc=%d\n",
|
||||
current_frame_size, max_frame_size, max_sb,
|
||||
bit_alloc[max_sb]);
|
||||
#endif
|
||||
if (max_sb < 0)
|
||||
break;
|
||||
|
||||
/* find alloc table entry (XXX: not optimal, should use
|
||||
pointer table) */
|
||||
alloc = s->alloc_table;
|
||||
for(i=0;i<max_sb;i++) {
|
||||
alloc += 1 << alloc[0];
|
||||
}
|
||||
|
||||
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
|
||||
/* nothing was coded for this band: add the necessary bits */
|
||||
incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
|
||||
incr += total_quant_bits[alloc[1]];
|
||||
} else {
|
||||
/* increments bit allocation */
|
||||
b = bit_alloc[max_ch][max_sb];
|
||||
incr = total_quant_bits[alloc[b + 1]] -
|
||||
total_quant_bits[alloc[b]];
|
||||
}
|
||||
|
||||
if (current_frame_size + incr <= max_frame_size) {
|
||||
/* can increase size */
|
||||
b = ++bit_alloc[max_ch][max_sb];
|
||||
current_frame_size += incr;
|
||||
/* decrease smr by the resolution we added */
|
||||
smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
|
||||
/* max allocation size reached ? */
|
||||
if (b == ((1 << alloc[0]) - 1))
|
||||
subband_status[max_ch][max_sb] = SB_NOMORE;
|
||||
else
|
||||
subband_status[max_ch][max_sb] = SB_ALLOCATED;
|
||||
} else {
|
||||
/* cannot increase the size of this subband */
|
||||
subband_status[max_ch][max_sb] = SB_NOMORE;
|
||||
}
|
||||
}
|
||||
*padding = max_frame_size - current_frame_size;
|
||||
assert(*padding >= 0);
|
||||
|
||||
#if 0
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
printf("%d ", bit_alloc[i]);
|
||||
}
|
||||
printf("\n");
|
||||
#endif
|
||||
}
|
||||
|
||||
/*
|
||||
* Output the mpeg audio layer 2 frame. Note how the code is small
|
||||
* compared to other encoders :-)
|
||||
*/
|
||||
static void encode_frame(MpegAudioContext *s,
|
||||
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
|
||||
int padding)
|
||||
{
|
||||
int i, j, k, l, bit_alloc_bits, b, ch;
|
||||
unsigned char *sf;
|
||||
int q[3];
|
||||
PutBitContext *p = &s->pb;
|
||||
|
||||
/* header */
|
||||
|
||||
put_bits(p, 12, 0xfff);
|
||||
put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
|
||||
put_bits(p, 2, 4-2); /* layer 2 */
|
||||
put_bits(p, 1, 1); /* no error protection */
|
||||
put_bits(p, 4, s->bitrate_index);
|
||||
put_bits(p, 2, s->freq_index);
|
||||
put_bits(p, 1, s->do_padding); /* use padding */
|
||||
put_bits(p, 1, 0); /* private_bit */
|
||||
put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
|
||||
put_bits(p, 2, 0); /* mode_ext */
|
||||
put_bits(p, 1, 0); /* no copyright */
|
||||
put_bits(p, 1, 1); /* original */
|
||||
put_bits(p, 2, 0); /* no emphasis */
|
||||
|
||||
/* bit allocation */
|
||||
j = 0;
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
bit_alloc_bits = s->alloc_table[j];
|
||||
for(ch=0;ch<s->nb_channels;ch++) {
|
||||
put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
|
||||
}
|
||||
j += 1 << bit_alloc_bits;
|
||||
}
|
||||
|
||||
/* scale codes */
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
for(ch=0;ch<s->nb_channels;ch++) {
|
||||
if (bit_alloc[ch][i])
|
||||
put_bits(p, 2, s->scale_code[ch][i]);
|
||||
}
|
||||
}
|
||||
|
||||
/* scale factors */
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
for(ch=0;ch<s->nb_channels;ch++) {
|
||||
if (bit_alloc[ch][i]) {
|
||||
sf = &s->scale_factors[ch][i][0];
|
||||
switch(s->scale_code[ch][i]) {
|
||||
case 0:
|
||||
put_bits(p, 6, sf[0]);
|
||||
put_bits(p, 6, sf[1]);
|
||||
put_bits(p, 6, sf[2]);
|
||||
break;
|
||||
case 3:
|
||||
case 1:
|
||||
put_bits(p, 6, sf[0]);
|
||||
put_bits(p, 6, sf[2]);
|
||||
break;
|
||||
case 2:
|
||||
put_bits(p, 6, sf[0]);
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* quantization & write sub band samples */
|
||||
|
||||
for(k=0;k<3;k++) {
|
||||
for(l=0;l<12;l+=3) {
|
||||
j = 0;
|
||||
for(i=0;i<s->sblimit;i++) {
|
||||
bit_alloc_bits = s->alloc_table[j];
|
||||
for(ch=0;ch<s->nb_channels;ch++) {
|
||||
b = bit_alloc[ch][i];
|
||||
if (b) {
|
||||
int qindex, steps, m, sample, bits;
|
||||
/* we encode 3 sub band samples of the same sub band at a time */
|
||||
qindex = s->alloc_table[j+b];
|
||||
steps = ff_mpa_quant_steps[qindex];
|
||||
for(m=0;m<3;m++) {
|
||||
sample = s->sb_samples[ch][k][l + m][i];
|
||||
/* divide by scale factor */
|
||||
#ifdef USE_FLOATS
|
||||
{
|
||||
float a;
|
||||
a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
|
||||
q[m] = (int)((a + 1.0) * steps * 0.5);
|
||||
}
|
||||
#else
|
||||
{
|
||||
int q1, e, shift, mult;
|
||||
e = s->scale_factors[ch][i][k];
|
||||
shift = scale_factor_shift[e];
|
||||
mult = scale_factor_mult[e];
|
||||
|
||||
/* normalize to P bits */
|
||||
if (shift < 0)
|
||||
q1 = sample << (-shift);
|
||||
else
|
||||
q1 = sample >> shift;
|
||||
q1 = (q1 * mult) >> P;
|
||||
q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
|
||||
}
|
||||
#endif
|
||||
if (q[m] >= steps)
|
||||
q[m] = steps - 1;
|
||||
assert(q[m] >= 0 && q[m] < steps);
|
||||
}
|
||||
bits = ff_mpa_quant_bits[qindex];
|
||||
if (bits < 0) {
|
||||
/* group the 3 values to save bits */
|
||||
put_bits(p, -bits,
|
||||
q[0] + steps * (q[1] + steps * q[2]));
|
||||
#if 0
|
||||
printf("%d: gr1 %d\n",
|
||||
i, q[0] + steps * (q[1] + steps * q[2]));
|
||||
#endif
|
||||
} else {
|
||||
#if 0
|
||||
printf("%d: gr3 %d %d %d\n",
|
||||
i, q[0], q[1], q[2]);
|
||||
#endif
|
||||
put_bits(p, bits, q[0]);
|
||||
put_bits(p, bits, q[1]);
|
||||
put_bits(p, bits, q[2]);
|
||||
}
|
||||
}
|
||||
}
|
||||
/* next subband in alloc table */
|
||||
j += 1 << bit_alloc_bits;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* padding */
|
||||
for(i=0;i<padding;i++)
|
||||
put_bits(p, 1, 0);
|
||||
|
||||
/* flush */
|
||||
flush_put_bits(p);
|
||||
}
|
||||
|
||||
static int MPA_encode_frame(AVCodecContext *avctx,
|
||||
unsigned char *frame, int buf_size, void *data)
|
||||
{
|
||||
MpegAudioContext *s = avctx->priv_data;
|
||||
short *samples = data;
|
||||
short smr[MPA_MAX_CHANNELS][SBLIMIT];
|
||||
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
|
||||
int padding, i;
|
||||
|
||||
for(i=0;i<s->nb_channels;i++) {
|
||||
filter(s, i, samples + i, s->nb_channels);
|
||||
}
|
||||
|
||||
for(i=0;i<s->nb_channels;i++) {
|
||||
compute_scale_factors(s->scale_code[i], s->scale_factors[i],
|
||||
s->sb_samples[i], s->sblimit);
|
||||
}
|
||||
for(i=0;i<s->nb_channels;i++) {
|
||||
psycho_acoustic_model(s, smr[i]);
|
||||
}
|
||||
compute_bit_allocation(s, smr, bit_alloc, &padding);
|
||||
|
||||
init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
|
||||
|
||||
encode_frame(s, bit_alloc, padding);
|
||||
|
||||
s->nb_samples += MPA_FRAME_SIZE;
|
||||
return pbBufPtr(&s->pb) - s->pb.buf;
|
||||
}
|
||||
|
||||
static int MPA_encode_close(AVCodecContext *avctx)
|
||||
{
|
||||
av_freep(&avctx->coded_frame);
|
||||
return 0;
|
||||
}
|
||||
|
||||
AVCodec mp2_encoder = {
|
||||
"mp2",
|
||||
CODEC_TYPE_AUDIO,
|
||||
CODEC_ID_MP2,
|
||||
sizeof(MpegAudioContext),
|
||||
MPA_encode_init,
|
||||
MPA_encode_frame,
|
||||
MPA_encode_close,
|
||||
NULL,
|
||||
};
|
||||
|
||||
#undef FIX
|
Loading…
Reference in New Issue
Block a user