diff options
Diffstat (limited to 'libavformat/audiointerleave.c')
-rw-r--r-- | libavformat/audiointerleave.c | 28 |
1 files changed, 18 insertions, 10 deletions
diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c index ba78d4e..b64010f 100644 --- a/libavformat/audiointerleave.c +++ b/libavformat/audiointerleave.c @@ -3,20 +3,20 @@ * * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com> * - * 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 */ @@ -34,7 +34,7 @@ void ff_audio_interleave_close(AVFormatContext *s) AudioInterleaveContext *aic = st->priv_data; if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) - av_fifo_free(aic->fifo); + av_fifo_freep(&aic->fifo); } } @@ -45,8 +45,12 @@ int ff_audio_interleave_init(AVFormatContext *s, int i; if (!samples_per_frame) - return -1; + return AVERROR(EINVAL); + if (!time_base.num) { + av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); + return AVERROR(EINVAL); + } for (i = 0; i < s->nb_streams; i++) { AVStream *st = s->streams[i]; AudioInterleaveContext *aic = st->priv_data; @@ -56,14 +60,15 @@ int ff_audio_interleave_init(AVFormatContext *s, av_get_bits_per_sample(st->codec->codec_id)) / 8; if (!aic->sample_size) { av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); - return -1; + return AVERROR(EINVAL); } aic->samples_per_frame = samples_per_frame; aic->samples = aic->samples_per_frame; aic->time_base = time_base; aic->fifo_size = 100* *aic->samples; - aic->fifo= av_fifo_alloc(100 * *aic->samples); + if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples))) + return AVERROR(ENOMEM); } } @@ -110,7 +115,7 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; if (new_size > aic->fifo_size) { if (av_fifo_realloc2(aic->fifo, new_size) < 0) - return -1; + return AVERROR(ENOMEM); aic->fifo_size = new_size; } av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL); @@ -128,9 +133,12 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt AVStream *st = s->streams[i]; if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { AVPacket new_pkt; - while (interleave_new_audio_packet(s, &new_pkt, i, flush)) + while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0) return ret; + } + if (ret < 0) + return ret; } } |