From 45ef963908f5ccc63161d4c3479ba8f2a56a7705 Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Mon, 4 Nov 2013 21:14:03 +0100 Subject: [PATCH] mpegaudioenc: Move some static tables to MpegAudioContext This reduces global state and the amount of globally visible tables. --- libavcodec/mpegaudioenc.c | 40 ++++++++++++++++++++++----------------- libavcodec/mpegaudiotab.h | 9 --------- 2 files changed, 23 insertions(+), 26 deletions(-) diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c index b7727ab950..a940c0d689 100644 --- a/libavcodec/mpegaudioenc.c +++ b/libavcodec/mpegaudioenc.c @@ -61,6 +61,11 @@ typedef struct MpegAudioContext { 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]; + float scale_factor_inv_table[64]; + unsigned short total_quant_bits[17]; /* total number of bits per allocation group */ } MpegAudioContext; static av_cold int MPA_encode_init(AVCodecContext *avctx) @@ -136,19 +141,19 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx) #if WFRAC_BITS != 16 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); #endif - filter_bank[i] = v; + s->filter_bank[i] = v; if ((i & 63) != 0) v = -v; if (i != 0) - filter_bank[512 - i] = v; + s->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; - scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); + s->scale_factor_table[i] = v; + s->scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); } for(i=0;i<128;i++) { v = i - 64; @@ -162,7 +167,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx) v = 3; else v = 4; - scale_diff_table[i] = v; + s->scale_diff_table[i] = v; } for(i=0;i<17;i++) { @@ -171,7 +176,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx) v = -v; else v = v * 3; - total_quant_bits[i] = 12 * v; + s->total_quant_bits[i] = 12 * v; } return 0; @@ -318,7 +323,7 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr) /* filter */ p = s->samples_buf[ch] + offset; - q = filter_bank; + q = s->filter_bank; /* maxsum = 23169 */ for(i=0;i<64;i++) { sum = p[0*64] * q[0*64]; @@ -352,7 +357,8 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr) s->samples_offset[ch] = offset; } -static void compute_scale_factors(unsigned char scale_code[SBLIMIT], +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) @@ -379,7 +385,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT], use at most 2 compares to find the index */ index = (21 - n) * 3 - 3; if (index >= 0) { - while (vmax <= scale_factor_table[index+1]) + while (vmax <= s->scale_factor_table[index+1]) index++; } else { index = 0; /* very unlikely case of overflow */ @@ -389,7 +395,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT], } av_dlog(NULL, "%2d:%d in=%x %x %d\n", - j, i, vmax, scale_factor_table[index], index); + j, i, vmax, s->scale_factor_table[index], index); /* store the scale factor */ assert(index >=0 && index <= 63); sf[i] = index; @@ -397,8 +403,8 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT], /* 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]; + 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) { @@ -548,12 +554,12 @@ static void compute_bit_allocation(MpegAudioContext *s, 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]]; + incr += s->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]]; + incr = s->total_quant_bits[alloc[b + 1]] - + s->total_quant_bits[alloc[b]]; } if (current_frame_size + incr <= max_frame_size) { @@ -665,7 +671,7 @@ static void encode_frame(MpegAudioContext *s, float a; sample = s->sb_samples[ch][k][l + m][i]; /* divide by scale factor */ - a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]]; + a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]]; q[m] = (int)((a + 1.0) * steps * 0.5); if (q[m] >= steps) q[m] = steps - 1; @@ -711,7 +717,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, } for(i=0;inb_channels;i++) { - compute_scale_factors(s->scale_code[i], s->scale_factors[i], + compute_scale_factors(s, s->scale_code[i], s->scale_factors[i], s->sb_samples[i], s->sblimit); } for(i=0;inb_channels;i++) { diff --git a/libavcodec/mpegaudiotab.h b/libavcodec/mpegaudiotab.h index 2addcb2630..d30ef1b752 100644 --- a/libavcodec/mpegaudiotab.h +++ b/libavcodec/mpegaudiotab.h @@ -79,15 +79,6 @@ static const int bitinv32[32] = { }; -static int16_t filter_bank[512]; - -static int scale_factor_table[64]; -static float scale_factor_inv_table[64]; -static unsigned char scale_diff_table[128]; - -/* total number of bits per allocation group */ -static unsigned short total_quant_bits[17]; - /* signal to noise ratio of each quantification step (could be computed from quant_steps[]). The values are dB multiplied by 10 */