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Diffstat (limited to 'libavcodec/mpegaudioenc.c')
-rw-r--r--libavcodec/mpegaudioenc.c20
1 files changed, 9 insertions, 11 deletions
diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index bc44aed12d..585ef44f87 100644
--- 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 Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -404,7 +404,7 @@ 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);
/* store the scale factor */
- assert(index >=0 && index <= 63);
+ av_assert2(index >=0 && index <= 63);
sf[i] = index;
}
@@ -466,7 +466,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
sf[1] = sf[2] = sf[0];
break;
default:
- assert(0); //cannot happen
+ av_assert2(0); //cannot happen
code = 0; /* kill warning */
}
@@ -586,7 +586,7 @@ static void compute_bit_allocation(MpegAudioContext *s,
}
}
*padding = max_frame_size - current_frame_size;
- assert(*padding >= 0);
+ av_assert0(*padding >= 0);
}
/*
@@ -701,7 +701,7 @@ static void encode_frame(MpegAudioContext *s,
#endif
if (q[m] >= steps)
q[m] = steps - 1;
- assert(q[m] >= 0 && q[m] < steps);
+ av_assert2(q[m] >= 0 && q[m] < steps);
}
bits = ff_mpa_quant_bits[qindex];
if (bits < 0) {
@@ -751,10 +751,8 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
compute_bit_allocation(s, smr, bit_alloc, &padding);
- if ((ret = ff_alloc_packet(avpkt, MPA_MAX_CODED_FRAME_SIZE))) {
- av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)))
return ret;
- }
init_put_bits(&s->pb, avpkt->data, avpkt->size);