ac3enc: Add codec-specific options for writing AC-3 metadata.

[ffmpeg] / libavcodec / mpegaudioenc.c

diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index ce1524ba9172f317c4fe3fd742578be1bd23c6f5..bd581cfaa4e8b6f000e756cd09254a636b7cd6e8 100644 (file)
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -2,20 +2,20 @@
  * The simplest mpeg audio layer 2 encoder
  * Copyright (c) 2000, 2001 Fabrice Bellard
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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 2.1 of the License, or (at your option) any later version.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav 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
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -122,7 +122,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
     s->sblimit = ff_mpa_sblimit_table[table];
     s->alloc_table = ff_mpa_alloc_tables[table];
 
-    dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
+    av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
             bitrate, freq, s->frame_size, table, s->frame_frac_incr);
 
     for(i=0;i<s->nb_channels;i++)
@@ -306,7 +306,7 @@ static void idct32(int *out, int *tab)
 
 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
 
-static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
+static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
 {
     short *p, *q;
     int sum, offset, i, j;
@@ -752,7 +752,7 @@ static int MPA_encode_frame(AVCodecContext *avctx,
                             unsigned char *frame, int buf_size, void *data)
 {
     MpegAudioContext *s = avctx->priv_data;
-    short *samples = data;
+    const short *samples = data;
     short smr[MPA_MAX_CHANNELS][SBLIMIT];
     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
     int padding, i;
@@ -783,7 +783,7 @@ static av_cold int MPA_encode_close(AVCodecContext *avctx)
     return 0;
 }
 
-AVCodec mp2_encoder = {
+AVCodec ff_mp2_encoder = {
     "mp2",
     AVMEDIA_TYPE_AUDIO,
     CODEC_ID_MP2,
@@ -792,7 +792,7 @@ AVCodec mp2_encoder = {
     MPA_encode_frame,
     MPA_encode_close,
     NULL,
-    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+    .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)"),
 };