mirror of
https://github.com/FFmpeg/FFmpeg.git
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219 lines
5.7 KiB
C
219 lines
5.7 KiB
C
/*
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* sndio grab interface
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* Copyright (c) 2010 Jacob Meuser
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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 <stdint.h>
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#include <sndio.h>
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#include "libavutil/internal.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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typedef struct SndioData {
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AVClass *class;
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struct sio_hdl *hdl;
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enum AVCodecID codec_id;
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int64_t hwpos;
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int64_t softpos;
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uint8_t *buffer;
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int bps;
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int buffer_size;
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int buffer_offset;
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int channels;
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int sample_rate;
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} SndioData;
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static inline void movecb(void *addr, int delta)
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{
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SndioData *s = addr;
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s->hwpos += delta * s->channels * s->bps;
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}
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static av_cold int sndio_open(AVFormatContext *s1, int is_output,
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const char *audio_device)
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{
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SndioData *s = s1->priv_data;
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struct sio_hdl *hdl;
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struct sio_par par;
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hdl = sio_open(audio_device, is_output ? SIO_PLAY : SIO_REC, 0);
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if (!hdl) {
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av_log(s1, AV_LOG_ERROR, "Could not open sndio device\n");
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return AVERROR(EIO);
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}
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sio_initpar(&par);
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par.bits = 16;
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par.sig = 1;
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par.le = SIO_LE_NATIVE;
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if (is_output)
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par.pchan = s->channels;
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else
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par.rchan = s->channels;
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par.rate = s->sample_rate;
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if (!sio_setpar(hdl, &par) || !sio_getpar(hdl, &par)) {
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av_log(s1, AV_LOG_ERROR, "Impossible to set sndio parameters, "
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"channels: %d sample rate: %d\n", s->channels, s->sample_rate);
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goto fail;
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}
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if (par.bits != 16 || par.sig != 1 ||
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(is_output && (par.pchan != s->channels)) ||
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(!is_output && (par.rchan != s->channels)) ||
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(par.rate != s->sample_rate)) {
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av_log(s1, AV_LOG_ERROR, "Could not set appropriate sndio parameters, "
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"channels: %d sample rate: %d\n", s->channels, s->sample_rate);
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goto fail;
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}
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s->buffer_size = par.round * par.bps *
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(is_output ? par.pchan : par.rchan);
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if (is_output) {
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s->buffer = av_malloc(s->buffer_size);
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if (!s->buffer) {
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av_log(s1, AV_LOG_ERROR, "Could not allocate buffer\n");
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goto fail;
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}
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}
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s->codec_id = par.le ? AV_CODEC_ID_PCM_S16LE : AV_CODEC_ID_PCM_S16BE;
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s->channels = is_output ? par.pchan : par.rchan;
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s->sample_rate = par.rate;
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s->bps = par.bps;
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sio_onmove(hdl, movecb, s);
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if (!sio_start(hdl)) {
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av_log(s1, AV_LOG_ERROR, "Could not start sndio\n");
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goto fail;
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}
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s->hdl = hdl;
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return 0;
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fail:
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av_freep(&s->buffer);
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if (hdl)
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sio_close(hdl);
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return AVERROR(EIO);
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}
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static av_cold int audio_read_header(AVFormatContext *s1)
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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 = 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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = s->codec_id;
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st->codecpar->sample_rate = s->sample_rate;
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st->codecpar->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_packet_unref(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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av_freep(&s->buffer);
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if (s->hdl)
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sio_close(s->hdl);
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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, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(SndioData, channels), AV_OPT_TYPE_INT, {.i64 = 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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