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https://github.com/FFmpeg/FFmpeg.git
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9d76cf0b18
* qatar/master: rtpdec: Templatize the code for different g726 bitrate variants rv40: move loop filter to rv34dsp context lavf: make av_set_pts_info private. rtpdec: Add support for G726 audio rtpdec: Add an init function that can do custom codec context initialization avconv: make copy_tb on by default. matroskadec: don't set codec timebase. rmdec: don't set codec timebase. avconv: compute next_pts from input packet duration when possible. lavf: estimate frame duration from r_frame_rate. avconv: update InputStream.pts in the streamcopy case. Conflicts: avconv.c libavdevice/alsa-audio-dec.c libavdevice/bktr.c libavdevice/fbdev.c libavdevice/libdc1394.c libavdevice/oss_audio.c libavdevice/v4l.c libavdevice/v4l2.c libavdevice/vfwcap.c libavdevice/x11grab.c libavformat/au.c libavformat/eacdata.c libavformat/flvdec.c libavformat/mpegts.c libavformat/mxfenc.c libavformat/rtpdec_g726.c libavformat/wtv.c libavformat/xmv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
119 lines
3.3 KiB
C
119 lines
3.3 KiB
C
/*
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* sndio play and grab interface
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* Copyright (c) 2010 Jacob Meuser
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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 <stdint.h>
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#include <sndio.h>
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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#include "libavutil/opt.h"
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#include "sndio_common.h"
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static av_cold int audio_read_header(AVFormatContext *s1,
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AVFormatParameters *ap)
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{
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SndioData *s = s1->priv_data;
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AVStream *st;
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int ret;
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st = avformat_new_stream(s1, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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ret = ff_sndio_open(s1, 0, s1->filename);
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if (ret < 0)
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return ret;
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/* take real parameters */
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = s->codec_id;
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st->codec->sample_rate = s->sample_rate;
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st->codec->channels = s->channels;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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}
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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SndioData *s = s1->priv_data;
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int64_t bdelay, cur_time;
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int ret;
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if ((ret = av_new_packet(pkt, s->buffer_size)) < 0)
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return ret;
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ret = sio_read(s->hdl, pkt->data, pkt->size);
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if (ret == 0 || sio_eof(s->hdl)) {
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av_free_packet(pkt);
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return AVERROR_EOF;
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}
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pkt->size = ret;
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s->softpos += ret;
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/* compute pts of the start of the packet */
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cur_time = av_gettime();
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bdelay = ret + s->hwpos - s->softpos;
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/* convert to pts */
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pkt->pts = cur_time - ((bdelay * 1000000) /
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(s->bps * s->channels * s->sample_rate));
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return 0;
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}
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static av_cold int audio_read_close(AVFormatContext *s1)
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{
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SndioData *s = s1->priv_data;
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ff_sndio_close(s);
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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(SndioData, sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(SndioData, channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass sndio_demuxer_class = {
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.class_name = "sndio indev",
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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_sndio_demuxer = {
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.name = "sndio",
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.long_name = NULL_IF_CONFIG_SMALL("sndio audio capture"),
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.priv_data_size = sizeof(SndioData),
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.read_header = audio_read_header,
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.read_packet = audio_read_packet,
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.read_close = audio_read_close,
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.flags = AVFMT_NOFILE,
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.priv_class = &sndio_demuxer_class,
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};
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