X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Faudiointerleave.c;h=e4cde9d0aab619a2db1aa6fe12e53fc0194c0bde;hb=2db953f84671997e936f91140ffb5143c1537844;hp=11f0093d559a6955c4e4b71dbc32ac3d1be92ad0;hpb=c957c8542676d092af1a812860c0b379738b7f01;p=ffmpeg diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c index 11f0093d559..e4cde9d0aab 100644 --- a/libavformat/audiointerleave.c +++ b/libavformat/audiointerleave.c @@ -3,24 +3,25 @@ * * Copyright (c) 2009 Baptiste Coudurier * - * 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 */ #include "libavutil/fifo.h" +#include "libavutil/mathematics.h" #include "avformat.h" #include "audiointerleave.h" #include "internal.h" @@ -32,7 +33,7 @@ void ff_audio_interleave_close(AVFormatContext *s) AVStream *st = s->streams[i]; AudioInterleaveContext *aic = st->priv_data; - if (st->codec->codec_type == CODEC_TYPE_AUDIO) + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) av_fifo_free(aic->fifo); } } @@ -50,7 +51,7 @@ int ff_audio_interleave_init(AVFormatContext *s, AVStream *st = s->streams[i]; AudioInterleaveContext *aic = st->priv_data; - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { aic->sample_size = (st->codec->channels * av_get_bits_per_sample(st->codec->codec_id)) / 8; if (!aic->sample_size) { @@ -69,8 +70,8 @@ int ff_audio_interleave_init(AVFormatContext *s, return 0; } -static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, - int stream_index, int flush) +static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, + int stream_index, int flush) { AVStream *st = s->streams[stream_index]; AudioInterleaveContext *aic = st->priv_data; @@ -80,7 +81,7 @@ static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, return 0; av_new_packet(pkt, size); - av_fifo_generic_read(aic->fifo, size, NULL, pkt->data); + av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); pkt->dts = pkt->pts = aic->dts; pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); @@ -103,7 +104,7 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt if (pkt) { AVStream *st = s->streams[pkt->stream_index]; AudioInterleaveContext *aic = st->priv_data; - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { 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) @@ -122,12 +123,12 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt for (i = 0; i < s->nb_streams; i++) { AVStream *st = s->streams[i]; - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { AVPacket new_pkt; - while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) + while (interleave_new_audio_packet(s, &new_pkt, i, flush)) ff_interleave_add_packet(s, &new_pkt, compare_ts); } } - return get_packet(s, out, pkt, flush); + return get_packet(s, out, NULL, flush); }