From 2456e28d9139fc6b2fee59dd058ad83a7e2f3416 Mon Sep 17 00:00:00 2001 From: Fabrice Bellard Date: Sat, 15 Sep 2001 22:42:25 +0000 Subject: merged code and tables between encoder and decoder Originally committed as revision 119 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/mpegaudio.c | 70 +++++++++++++++++++++++++++++++------------------- 1 file changed, 43 insertions(+), 27 deletions(-) (limited to 'libavcodec/mpegaudio.c') diff --git a/libavcodec/mpegaudio.c b/libavcodec/mpegaudio.c index ec8c82de3e..72661bbc0d 100644 --- a/libavcodec/mpegaudio.c +++ b/libavcodec/mpegaudio.c @@ -20,14 +20,36 @@ #include #include "mpegaudio.h" +#define DCT_BITS 14 /* number of bits for the DCT */ +#define MUL(a,b) (((a) * (b)) >> DCT_BITS) +#define FIX(a) ((int)((a) * (1 << DCT_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 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 -#define MPA_STEREO 0 -#define MPA_JSTEREO 1 -#define MPA_DUAL 2 -#define MPA_MONO 3 - #include "mpegaudiotab.h" int MPA_encode_init(AVCodecContext *avctx) @@ -36,7 +58,7 @@ int MPA_encode_init(AVCodecContext *avctx) int freq = avctx->sample_rate; int bitrate = avctx->bit_rate; int channels = avctx->channels; - int i, v, table, ch_bitrate; + int i, v, table; float a; if (channels > 2) @@ -51,9 +73,9 @@ int MPA_encode_init(AVCodecContext *avctx) /* encoding freq */ s->lsf = 0; for(i=0;i<3;i++) { - if (freq_tab[i] == freq) + if (mpa_freq_tab[i] == freq) break; - if ((freq_tab[i] / 2) == freq) { + if ((mpa_freq_tab[i] / 2) == freq) { s->lsf = 1; break; } @@ -64,7 +86,7 @@ int MPA_encode_init(AVCodecContext *avctx) /* encoding bitrate & frequency */ for(i=0;i<15;i++) { - if (bitrate_tab[1-s->lsf][i] == bitrate) + if (mpa_bitrate_tab[s->lsf][1][i] == bitrate) break; } if (i == 15) @@ -81,20 +103,8 @@ int MPA_encode_init(AVCodecContext *avctx) s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); /* select the right allocation table */ - ch_bitrate = bitrate / s->nb_channels; - if (!s->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; - } + table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf); + /* number of used subbands */ s->sblimit = sblimit_table[table]; s->alloc_table = alloc_tables[table]; @@ -107,10 +117,16 @@ int MPA_encode_init(AVCodecContext *avctx) for(i=0;inb_channels;i++) s->samples_offset[i] = 0; - for(i=0;i<512;i++) { - float a = enwindow[i] * 32768.0 * 16.0; - filter_bank[i] = (int)(a); + for(i=0;i<257;i++) { + int v; + v = (mpa_enwindow[i] + 2) >> 2; + 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) @@ -151,7 +167,7 @@ int MPA_encode_init(AVCodecContext *avctx) return 0; } -/* 32 point floating point IDCT */ +/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ static void idct32(int *out, int *tab, int sblimit, int left_shift) { int i, j; -- cgit v1.2.3