mirror of https://git.ffmpeg.org/ffmpeg.git
134 lines
4.0 KiB
C
134 lines
4.0 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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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/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 {
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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_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 * 32;
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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_free_packet(pkt);
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return ret < 0 ? ret : AVERROR(EIO);
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}
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pkt->size = ret;
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pkt->duration = ret / GSM_BLOCK_SIZE;
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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, AVFormatParameters *ap)
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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->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = s->iformat->value;
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st->codec->channels = 1;
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st->codec->sample_rate = c->sample_rate;
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st->codec->block_align = GSM_BLOCK_SIZE;
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st->codec->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 int gsm_read_seek2(AVFormatContext *s, int stream_index, int64_t min_ts,
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int64_t ts, int64_t max_ts, int flags)
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{
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GSMDemuxerContext *c = s->priv_data;
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/* convert timestamps to file positions */
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if (!(flags & AVSEEK_FLAG_BYTE)) {
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if (stream_index < 0) {
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AVRational bitrate_q = { GSM_BLOCK_SAMPLES, c->sample_rate * GSM_BLOCK_SIZE };
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ts = av_rescale_q(ts, AV_TIME_BASE_Q, bitrate_q);
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min_ts = av_rescale_q(min_ts, AV_TIME_BASE_Q, bitrate_q);
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max_ts = av_rescale_q(max_ts, AV_TIME_BASE_Q, bitrate_q);
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} else {
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ts *= GSM_BLOCK_SIZE;
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min_ts *= GSM_BLOCK_SIZE;
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max_ts *= GSM_BLOCK_SIZE;
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}
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}
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/* round to nearest block boundary */
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ts = (ts + GSM_BLOCK_SIZE / 2) / GSM_BLOCK_SIZE * GSM_BLOCK_SIZE;
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ts = FFMAX(0, ts);
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/* handle min/max */
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while (ts < min_ts)
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ts += GSM_BLOCK_SIZE;
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while (ts > max_ts)
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ts -= GSM_BLOCK_SIZE;
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if (ts < min_ts || ts > max_ts)
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return -1;
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return avio_seek(s->pb, ts, SEEK_SET);
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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, {.dbl = 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 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_header = gsm_read_header,
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.read_packet = gsm_read_packet,
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.read_seek2 = gsm_read_seek2,
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.extensions = "gsm",
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.value = CODEC_ID_GSM,
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.priv_class = &class,
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};
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