Merge commit '16381923fb7b9087ce559fb1cd3594469ac6788b'

[ffmpeg] / libavcodec / mpegaudioenc.c

diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index 63c8460c323625d34a1d91ebdc1475b6b2bec982..005401946eda5f312b33a836fc7d3c537932da8f 100644 (file)
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -33,8 +33,13 @@
 #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) */
@@ -59,14 +64,18 @@ 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];
+#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;
 
-/* define it to use floats in quantization (I don't like floats !) */
-#define USE_FLOATS
-
-#include "mpegaudiodata.h"
-#include "mpegaudiotab.h"
-
 static av_cold int MPA_encode_init(AVCodecContext *avctx)
 {
     MpegAudioContext *s = avctx->priv_data;
@@ -140,24 +149,24 @@ 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)(exp2((3 - i) / 3.0) * (1 << 20));
         if (v <= 0)
             v = 1;
-        scale_factor_table[i] = v;
+        s->scale_factor_table[i] = v;
 #ifdef USE_FLOATS
-        scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
+        s->scale_factor_inv_table[i] = exp2(-(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) * exp2((i % 3) / 3.0);
+        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++) {
@@ -172,7 +181,7 @@ 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++) {
@@ -181,7 +190,7 @@ 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;
@@ -328,7 +337,7 @@ 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];
@@ -362,7 +371,8 @@ 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)
@@ -389,7 +399,7 @@ 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 */
@@ -399,7 +409,7 @@ 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 */
             av_assert2(index >=0 && index <= 63);
             sf[i] = index;
@@ -407,8 +417,8 @@ 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) {
@@ -558,12 +568,12 @@ 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) {
@@ -677,15 +687,15 @@ static void encode_frame(MpegAudioContext *s,
 #ifdef USE_FLOATS
                             {
                                 float a;
-                                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);
                             }
 #else
                             {
                                 int q1, e, shift, mult;
                                 e = s->scale_factors[ch][i][k];
-                                shift = scale_factor_shift[e];
-                                mult = scale_factor_mult[e];
+                                shift = s->scale_factor_shift[e];
+                                mult = s->scale_factor_mult[e];
 
                                 /* normalize to P bits */
                                 if (shift < 0)
@@ -740,7 +750,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     }
 
     for(i=0;i<s->nb_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;i<s->nb_channels;i++) {