Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavcodec / mpegaudioenc.c

diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index a5859837ef64e04c085825b7bd6aefa881f56280..cc814091ec8d0b90f6579775674ed91851375ff1 100644 (file)
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -25,6 +25,7 @@
  */
 
 #include "avcodec.h"
+#include "internal.h"
 #include "put_bits.h"
 
 #define FRAC_BITS   15   /* fractional bits for sb_samples and dct */
@@ -315,8 +316,6 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
     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++) {
@@ -360,8 +359,6 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
         }
     }
     s->samples_offset[ch] = offset;
-
-    //    print_pow(s->sb_samples, 1152);
 }
 
 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
@@ -400,10 +397,8 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
                 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
+            av_dlog(NULL, "%2d:%d in=%x %x %d\n",
+                    j, i, vmax, scale_factor_table[index], index);
             /* store the scale factor */
             assert(index >=0 && index <= 63);
             sf[i] = index;
@@ -471,10 +466,8 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
             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
+        av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
+                sf[0], sf[1], sf[2], d1, d2, code);
         scale_code[j] = code;
         sf += 3;
     }
@@ -767,16 +760,21 @@ static av_cold int MPA_encode_close(AVCodecContext *avctx)
     return 0;
 }
 
+static const AVCodecDefault mp2_defaults[] = {
+    { "b",    "128k" },
+    { NULL },
+};
+
 AVCodec ff_mp2_encoder = {
-    "mp2",
-    AVMEDIA_TYPE_AUDIO,
-    CODEC_ID_MP2,
-    sizeof(MpegAudioContext),
-    MPA_encode_init,
-    MPA_encode_frame,
-    MPA_encode_close,
-    NULL,
+    .name           = "mp2",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_MP2,
+    .priv_data_size = sizeof(MpegAudioContext),
+    .init           = MPA_encode_init,
+    .encode         = MPA_encode_frame,
+    .close          = MPA_encode_close,
     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
     .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},
     .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
+    .defaults       = mp2_defaults,
 };