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https://git.ffmpeg.org/ffmpeg.git
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6f69f7a8bf
* commit '9200514ad8717c63f82101dc394f4378854325bf':
lavf: replace AVStream.codec with AVStream.codecpar
This has been a HUGE effort from:
- Derek Buitenhuis <derek.buitenhuis@gmail.com>
- Hendrik Leppkes <h.leppkes@gmail.com>
- wm4 <nfxjfg@googlemail.com>
- Clément Bœsch <clement@stupeflix.com>
- James Almer <jamrial@gmail.com>
- Michael Niedermayer <michael@niedermayer.cc>
- Rostislav Pehlivanov <atomnuker@gmail.com>
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
119 lines
3.3 KiB
C
119 lines
3.3 KiB
C
/*
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* RAW GSM demuxer
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* Copyright (c) 2011 Justin Ruggles
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "avformat.h"
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#include "internal.h"
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#define GSM_BLOCK_SIZE 33
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#define GSM_BLOCK_SAMPLES 160
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#define GSM_SAMPLE_RATE 8000
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typedef struct GSMDemuxerContext {
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AVClass *class;
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int sample_rate;
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} GSMDemuxerContext;
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static int gsm_probe(AVProbeData *p)
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{
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int valid = 0, invalid = 0;
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uint8_t *b = p->buf;
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while (b < p->buf + p->buf_size - 32) {
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if ((*b & 0xf0) == 0xd0) {
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valid++;
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} else {
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invalid++;
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}
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b += 33;
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}
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if (valid >> 5 > invalid)
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return AVPROBE_SCORE_EXTENSION + 1;
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return 0;
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}
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static int gsm_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret, size;
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size = GSM_BLOCK_SIZE;
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pkt->pos = avio_tell(s->pb);
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pkt->stream_index = 0;
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ret = av_get_packet(s->pb, pkt, size);
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if (ret < GSM_BLOCK_SIZE) {
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av_packet_unref(pkt);
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return ret < 0 ? ret : AVERROR(EIO);
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}
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pkt->duration = 1;
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pkt->pts = pkt->pos / GSM_BLOCK_SIZE;
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return 0;
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}
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static int gsm_read_header(AVFormatContext *s)
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{
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GSMDemuxerContext *c = s->priv_data;
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AVStream *st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = s->iformat->raw_codec_id;
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st->codecpar->channels = 1;
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st->codecpar->channel_layout = AV_CH_LAYOUT_MONO;
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st->codecpar->sample_rate = c->sample_rate;
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st->codecpar->bit_rate = GSM_BLOCK_SIZE * 8 * c->sample_rate / GSM_BLOCK_SAMPLES;
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avpriv_set_pts_info(st, 64, GSM_BLOCK_SAMPLES, GSM_SAMPLE_RATE);
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return 0;
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}
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static const AVOption options[] = {
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{ "sample_rate", "", offsetof(GSMDemuxerContext, sample_rate),
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AV_OPT_TYPE_INT, {.i64 = GSM_SAMPLE_RATE}, 1, INT_MAX / GSM_BLOCK_SIZE,
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AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass gsm_class = {
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.class_name = "gsm demuxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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AVInputFormat ff_gsm_demuxer = {
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.name = "gsm",
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.long_name = NULL_IF_CONFIG_SMALL("raw GSM"),
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.priv_data_size = sizeof(GSMDemuxerContext),
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.read_probe = gsm_probe,
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.read_header = gsm_read_header,
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.read_packet = gsm_read_packet,
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.flags = AVFMT_GENERIC_INDEX,
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.extensions = "gsm",
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.raw_codec_id = AV_CODEC_ID_GSM,
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.priv_class = &gsm_class,
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};
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