Add a Makefile variable for the old scaler that gets enabled when the new

[ffmpeg] / libavcodec / ac3enc.c

diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c
index 5243b4fbc461742bb63a46199d8c79770d2fb3cd..434869cee910f31567c4356f35978b4749690c7c 100644 (file)
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -1,5 +1,5 @@
 /*
- * The simplest AC3 encoder
+ * The simplest AC-3 encoder
  * Copyright (c) 2000 Fabrice Bellard.
  *
  * This file is part of FFmpeg.
@@ -21,13 +21,13 @@
 
 /**
  * @file ac3enc.c
- * The simplest AC3 encoder.
+ * The simplest AC-3 encoder.
  */
 //#define DEBUG
 //#define DEBUG_BITALLOC
+#include "libavutil/crc.h"
 #include "avcodec.h"
 #include "bitstream.h"
-#include "crc.h"
 #include "ac3.h"
 
 typedef struct AC3EncodeContext {
@@ -64,7 +64,6 @@ typedef struct AC3EncodeContext {
 
 static int16_t costab[64];
 static int16_t sintab[64];
-static int16_t fft_rev[512];
 static int16_t xcos1[128];
 static int16_t xsin1[128];
 
@@ -74,8 +73,6 @@ static int16_t xsin1[128];
 /* new exponents are sent if their Norm 1 exceed this number */
 #define EXP_DIFF_THRESHOLD 1000
 
-static void fft_init(int ln);
-
 static inline int16_t fix15(float a)
 {
     int v;
@@ -93,7 +90,7 @@ typedef struct IComplex {
 
 static void fft_init(int ln)
 {
-    int i, j, m, n;
+    int i, n;
     float alpha;
 
     n = 1 << ln;
@@ -103,14 +100,6 @@ static void fft_init(int ln)
         costab[i] = fix15(cos(alpha));
         sintab[i] = fix15(sin(alpha));
     }
-
-    for(i=0;i<n;i++) {
-        m=0;
-        for(j=0;j<ln;j++) {
-            m |= ((i >> j) & 1) << (ln-j-1);
-        }
-        fft_rev[i]=m;
-    }
 }
 
 /* butter fly op */
@@ -148,8 +137,7 @@ static void fft(IComplex *z, int ln)
 
     /* reverse */
     for(j=0;j<np;j++) {
-        int k;
-        k = fft_rev[j];
+        int k = ff_reverse[j] >> (8 - ln);
         if (k < j)
             FFSWAP(IComplex, z[k], z[j]);
     }
@@ -475,7 +463,8 @@ static int bit_alloc(AC3EncodeContext *s,
         for(ch=0;ch<s->nb_all_channels;ch++) {
             ff_ac3_bit_alloc_calc_bap(mask[i][ch], psd[i][ch], 0,
                                       s->nb_coefs[ch], snr_offset,
-                                      s->bit_alloc.floor, bap[i][ch]);
+                                      s->bit_alloc.floor, ff_ac3_bap_tab,
+                                      bap[i][ch]);
             frame_bits += compute_mantissa_size(s, bap[i][ch],
                                                  s->nb_coefs[ch]);
         }
@@ -501,7 +490,7 @@ static int compute_bit_allocation(AC3EncodeContext *s,
     uint8_t bap1[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
     int16_t psd[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
     int16_t mask[NB_BLOCKS][AC3_MAX_CHANNELS][50];
-    static int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
+    static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
 
     /* init default parameters */
     s->slow_decay_code = 2;
@@ -620,7 +609,7 @@ static int compute_bit_allocation(AC3EncodeContext *s,
     return 0;
 }
 
-static int AC3_encode_init(AVCodecContext *avctx)
+static av_cold int AC3_encode_init(AVCodecContext *avctx)
 {
     int freq = avctx->sample_rate;
     int bitrate = avctx->bit_rate;
@@ -688,7 +677,7 @@ static int AC3_encode_init(AVCodecContext *avctx)
     } else {
         /* use default bandwidth setting */
         /* XXX: should compute the bandwidth according to the frame
-           size, so that we avoid anoying high freq artefacts */
+           size, so that we avoid annoying high frequency artifacts */
         bw_code = 50;
     }
     for(ch=0;ch<s->nb_channels;ch++) {
@@ -716,7 +705,7 @@ static int AC3_encode_init(AVCodecContext *avctx)
     return 0;
 }
 
-/* output the AC3 frame header */
+/* output the AC-3 frame header */
 static void output_frame_header(AC3EncodeContext *s, unsigned char *frame)
 {
     init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE);
@@ -783,7 +772,7 @@ static inline int asym_quant(int c, int e, int qbits)
     return v & ((1 << qbits)-1);
 }
 
-/* Output one audio block. There are NB_BLOCKS audio blocks in one AC3
+/* Output one audio block. There are NB_BLOCKS audio blocks in one AC-3
    frame */
 static void output_audio_block(AC3EncodeContext *s,
                                uint8_t exp_strategy[AC3_MAX_CHANNELS],
@@ -1267,7 +1256,7 @@ static int AC3_encode_frame(AVCodecContext *avctx,
     return output_frame_end(s);
 }
 
-static int AC3_encode_close(AVCodecContext *avctx)
+static av_cold int AC3_encode_close(AVCodecContext *avctx)
 {
     av_freep(&avctx->coded_frame);
     return 0;
@@ -1375,4 +1364,6 @@ AVCodec ac3_encoder = {
     AC3_encode_frame,
     AC3_encode_close,
     NULL,
+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
 };