diff options
Diffstat (limited to 'libavcodec/mpegaudioenc.c')
-rw-r--r-- | libavcodec/mpegaudioenc.c | 20 |
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); |