QDM2 compatible decoder

[libav.git] / libavcodec / mpegaudio.c

diff --git a/libavcodec/mpegaudio.c b/libavcodec/mpegaudio.c
index e8ec337..7a0b0a3 100644 (file)
--- a/libavcodec/mpegaudio.c
+++ b/libavcodec/mpegaudio.c
@@ -1,30 +1,34 @@
 /*
  * The simplest mpeg audio layer 2 encoder
- * Copyright (c) 2000, 2001 Gerard Lantau.
+ * Copyright (c) 2000, 2001 Fabrice Bellard.
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
  *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  */
+ 
+/**
+ * @file mpegaudio.c
+ * The simplest mpeg audio layer 2 encoder.
+ */
+ 
 #include "avcodec.h"
-#include <math.h>
+#include "bitstream.h"
 #include "mpegaudio.h"
 
 /* currently, cannot change these constants (need to modify
    quantization stage) */
-#define FRAC_BITS 15
-#define WFRAC_BITS  14
-#define MUL(a,b) (((INT64)(a) * (INT64)(b)) >> FRAC_BITS)
+#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
 #define FIX(a)   ((int)((a) * (1 << FRAC_BITS)))
 
 #define SAMPLES_BUF_SIZE 4096
@@ -37,7 +41,7 @@ typedef struct MpegAudioContext {
     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 */
+    int64_t 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 */
@@ -55,7 +59,7 @@ typedef struct MpegAudioContext {
 
 #include "mpegaudiotab.h"
 
-int MPA_encode_init(AVCodecContext *avctx)
+static int MPA_encode_init(AVCodecContext *avctx)
 {
     MpegAudioContext *s = avctx->priv_data;
     int freq = avctx->sample_rate;
@@ -71,7 +75,6 @@ int MPA_encode_init(AVCodecContext *avctx)
     s->freq = freq;
     s->bit_rate = bitrate * 1000;
     avctx->frame_size = MPA_FRAME_SIZE;
-    avctx->key_frame = 1; /* always key frame */
 
     /* encoding freq */
     s->lsf = 0;
@@ -83,8 +86,10 @@ int MPA_encode_init(AVCodecContext *avctx)
             break;
         }
     }
-    if (i == 3)
+    if (i == 3){
+        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
         return -1;
+    }
     s->freq_index = i;
 
     /* encoding bitrate & frequency */
@@ -92,8 +97,10 @@ int MPA_encode_init(AVCodecContext *avctx)
         if (mpa_bitrate_tab[s->lsf][1][i] == bitrate) 
             break;
     }
-    if (i == 15)
+    if (i == 15){
+        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
         return -1;
+    }
     s->bitrate_index = i;
 
     /* compute total header size & pad bit */
@@ -113,7 +120,7 @@ int MPA_encode_init(AVCodecContext *avctx)
     s->alloc_table = alloc_tables[table];
 
 #ifdef DEBUG
-    printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 
+    av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 
            bitrate, freq, s->frame_size, table, s->frame_frac_incr);
 #endif
 
@@ -170,6 +177,9 @@ int MPA_encode_init(AVCodecContext *avctx)
         total_quant_bits[i] = 12 * v;
     }
 
+    avctx->coded_frame= avcodec_alloc_frame();
+    avctx->coded_frame->key_frame= 1;
+
     return 0;
 }
 
@@ -455,7 +465,8 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
             sf[1] = sf[2] = sf[0];
             break;
         default:
-            abort();
+            assert(0); //cant happen
+            code = 0;           /* kill warning */
         }
         
 #if 0
@@ -736,8 +747,8 @@ static void encode_frame(MpegAudioContext *s,
     flush_put_bits(p);
 }
 
-int MPA_encode_frame(AVCodecContext *avctx,
-                     unsigned char *frame, int buf_size, void *data)
+static int MPA_encode_frame(AVCodecContext *avctx,
+                           unsigned char *frame, int buf_size, void *data)
 {
     MpegAudioContext *s = avctx->priv_data;
     short *samples = data;
@@ -758,15 +769,21 @@ int MPA_encode_frame(AVCodecContext *avctx,
     }
     compute_bit_allocation(s, smr, bit_alloc, &padding);
 
-    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
+    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
 
     encode_frame(s, bit_alloc, padding);
     
     s->nb_samples += MPA_FRAME_SIZE;
-    return s->pb.buf_ptr - s->pb.buf;
+    return pbBufPtr(&s->pb) - s->pb.buf;
 }
 
+static int MPA_encode_close(AVCodecContext *avctx)
+{
+    av_freep(&avctx->coded_frame);
+    return 0;
+}
 
+#ifdef CONFIG_MP2_ENCODER
 AVCodec mp2_encoder = {
     "mp2",
     CODEC_TYPE_AUDIO,
@@ -774,5 +791,9 @@ AVCodec mp2_encoder = {
     sizeof(MpegAudioContext),
     MPA_encode_init,
     MPA_encode_frame,
+    MPA_encode_close,
     NULL,
 };
+#endif // CONFIG_MP2_ENCODER
+
+#undef FIX