mirror of
https://github.com/librempeg/librempeg
synced 2024-11-23 11:39:47 +00:00
338 lines
8.8 KiB
C
338 lines
8.8 KiB
C
/*
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* Linux audio play and grab interface
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* Copyright (c) 2000, 2001 Fabrice Bellard
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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 "config.h"
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#include <stdlib.h>
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#include <stdio.h>
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#include <stdint.h>
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#include <string.h>
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#include <errno.h>
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#if HAVE_SOUNDCARD_H
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#include <soundcard.h>
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#else
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#include <sys/soundcard.h>
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#endif
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#include <unistd.h>
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include "libavutil/internal.h"
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#include "libavutil/log.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "libavcodec/avcodec.h"
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#include "avdevice.h"
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#include "libavformat/internal.h"
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#define AUDIO_BLOCK_SIZE 4096
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typedef struct {
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AVClass *class;
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int fd;
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int sample_rate;
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int channels;
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int frame_size; /* in bytes ! */
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enum AVCodecID codec_id;
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unsigned int flip_left : 1;
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uint8_t buffer[AUDIO_BLOCK_SIZE];
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int buffer_ptr;
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} AudioData;
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static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
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{
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AudioData *s = s1->priv_data;
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int audio_fd;
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int tmp, err;
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char *flip = getenv("AUDIO_FLIP_LEFT");
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if (is_output)
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audio_fd = avpriv_open(audio_device, O_WRONLY);
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else
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audio_fd = avpriv_open(audio_device, O_RDONLY);
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if (audio_fd < 0) {
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av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
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return AVERROR(EIO);
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}
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if (flip && *flip == '1') {
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s->flip_left = 1;
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}
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/* non blocking mode */
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if (!is_output) {
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if (fcntl(audio_fd, F_SETFL, O_NONBLOCK) < 0) {
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av_log(s1, AV_LOG_WARNING, "%s: Could not enable non block mode (%s)\n", audio_device, strerror(errno));
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}
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}
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s->frame_size = AUDIO_BLOCK_SIZE;
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/* select format : favour native format */
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err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
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#if HAVE_BIGENDIAN
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if (tmp & AFMT_S16_BE) {
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tmp = AFMT_S16_BE;
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} else if (tmp & AFMT_S16_LE) {
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tmp = AFMT_S16_LE;
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} else {
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tmp = 0;
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}
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#else
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if (tmp & AFMT_S16_LE) {
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tmp = AFMT_S16_LE;
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} else if (tmp & AFMT_S16_BE) {
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tmp = AFMT_S16_BE;
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} else {
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tmp = 0;
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}
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#endif
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switch(tmp) {
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case AFMT_S16_LE:
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s->codec_id = AV_CODEC_ID_PCM_S16LE;
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break;
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case AFMT_S16_BE:
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s->codec_id = AV_CODEC_ID_PCM_S16BE;
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break;
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default:
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av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
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close(audio_fd);
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return AVERROR(EIO);
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}
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err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
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if (err < 0) {
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av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
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goto fail;
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}
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tmp = (s->channels == 2);
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err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
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if (err < 0) {
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av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
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goto fail;
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}
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tmp = s->sample_rate;
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err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
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if (err < 0) {
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av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
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goto fail;
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}
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s->sample_rate = tmp; /* store real sample rate */
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s->fd = audio_fd;
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return 0;
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fail:
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close(audio_fd);
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return AVERROR(EIO);
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}
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static int audio_close(AudioData *s)
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{
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close(s->fd);
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return 0;
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}
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/* sound output support */
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static int audio_write_header(AVFormatContext *s1)
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{
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AudioData *s = s1->priv_data;
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AVStream *st;
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int ret;
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st = s1->streams[0];
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s->sample_rate = st->codec->sample_rate;
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s->channels = st->codec->channels;
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ret = audio_open(s1, 1, s1->filename);
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if (ret < 0) {
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return AVERROR(EIO);
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} else {
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return 0;
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}
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}
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static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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AudioData *s = s1->priv_data;
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int len, ret;
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int size= pkt->size;
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uint8_t *buf= pkt->data;
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while (size > 0) {
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len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
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memcpy(s->buffer + s->buffer_ptr, buf, len);
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s->buffer_ptr += len;
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if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
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for(;;) {
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ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
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if (ret > 0)
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break;
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if (ret < 0 && (errno != EAGAIN && errno != EINTR))
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return AVERROR(EIO);
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}
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s->buffer_ptr = 0;
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}
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buf += len;
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size -= len;
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}
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return 0;
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}
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static int audio_write_trailer(AVFormatContext *s1)
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{
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AudioData *s = s1->priv_data;
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audio_close(s);
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return 0;
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}
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/* grab support */
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static int audio_read_header(AVFormatContext *s1)
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{
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AudioData *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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}
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ret = audio_open(s1, 0, s1->filename);
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if (ret < 0) {
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return AVERROR(EIO);
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}
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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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AudioData *s = s1->priv_data;
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int ret, bdelay;
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int64_t cur_time;
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struct audio_buf_info abufi;
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if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
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return ret;
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ret = read(s->fd, pkt->data, pkt->size);
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if (ret <= 0){
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av_free_packet(pkt);
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pkt->size = 0;
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if (ret<0) return AVERROR(errno);
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else return AVERROR_EOF;
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}
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pkt->size = 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;
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if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
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bdelay += abufi.bytes;
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}
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/* subtract time represented by the number of bytes in the audio fifo */
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cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
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/* convert to wanted units */
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pkt->pts = cur_time;
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if (s->flip_left && s->channels == 2) {
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int i;
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short *p = (short *) pkt->data;
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for (i = 0; i < ret; i += 4) {
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*p = ~*p;
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p += 2;
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}
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}
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return 0;
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}
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static int audio_read_close(AVFormatContext *s1)
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{
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AudioData *s = s1->priv_data;
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audio_close(s);
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return 0;
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}
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#if CONFIG_OSS_INDEV
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static const AVOption options[] = {
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{ "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(AudioData, 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 oss_demuxer_class = {
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.class_name = "OSS 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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.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
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};
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AVInputFormat ff_oss_demuxer = {
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.name = "oss",
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.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
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.priv_data_size = sizeof(AudioData),
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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 = &oss_demuxer_class,
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};
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#endif
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#if CONFIG_OSS_OUTDEV
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static const AVClass oss_muxer_class = {
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.class_name = "OSS muxer",
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.item_name = av_default_item_name,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
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};
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AVOutputFormat ff_oss_muxer = {
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.name = "oss",
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.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
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.priv_data_size = sizeof(AudioData),
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/* XXX: we make the assumption that the soundcard accepts this format */
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/* XXX: find better solution with "preinit" method, needed also in
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other formats */
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.audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
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.video_codec = AV_CODEC_ID_NONE,
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.write_header = audio_write_header,
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.write_packet = audio_write_packet,
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.write_trailer = audio_write_trailer,
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.flags = AVFMT_NOFILE,
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.priv_class = &oss_muxer_class,
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};
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#endif
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