diff options
Diffstat (limited to 'libav/audio.c')
| -rw-r--r-- | libav/audio.c | 179 |
1 files changed, 179 insertions, 0 deletions
diff --git a/libav/audio.c b/libav/audio.c new file mode 100644 index 0000000000..df5d88eaf6 --- /dev/null +++ b/libav/audio.c @@ -0,0 +1,179 @@ +/* + * Linux audio play and grab interface + * Copyright (c) 2000, 2001 Gerard Lantau. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ +#include <stdlib.h> +#include <stdio.h> +#include <string.h> +#include <linux/soundcard.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/ioctl.h> +#include <sys/mman.h> +#include <errno.h> +#include <sys/time.h> + +#include "avformat.h" + +const char *audio_device = "/dev/dsp"; + +typedef struct { + int fd; + int rate; + int channels; +} AudioData; + +#define AUDIO_BLOCK_SIZE 4096 + +/* audio read support */ + +static int audio_read(URLContext *h, UINT8 *buf, int size) +{ + AudioData *s = h->priv_data; + int ret; + + ret = read(s->fd, buf, size); + if (ret < 0) + return -errno; + else + return ret; +} + +static int audio_write(URLContext *h, UINT8 *buf, int size) +{ + AudioData *s = h->priv_data; + int ret; + + ret = write(s->fd, buf, size); + if (ret < 0) + return -errno; + else + return ret; +} + +static int audio_get_format(URLContext *h, URLFormat *f) +{ + AudioData *s = h->priv_data; + + strcpy(f->format_name, "pcm"); + f->sample_rate = s->rate; + f->channels = s->channels; + return 0; +} + +/* URI syntax: 'audio:[rate[,channels]]' + default: rate=44100, channels=2 + */ +static int audio_open(URLContext *h, const char *uri, int flags) +{ + AudioData *s; + const char *p; + int freq, channels, audio_fd; + int tmp, err; + + h->is_streamed = 1; + h->packet_size = AUDIO_BLOCK_SIZE; + + s = malloc(sizeof(AudioData)); + if (!s) + return -ENOMEM; + h->priv_data = s; + + /* extract parameters */ + p = uri; + strstart(p, "audio:", &p); + freq = strtol(p, (char **)&p, 0); + if (freq <= 0) + freq = 44100; + if (*p == ',') + p++; + channels = strtol(p, (char **)&p, 0); + if (channels <= 0) + channels = 2; + s->rate = freq; + s->channels = channels; + + /* open linux audio device */ + if (flags & URL_WRONLY) + audio_fd = open(audio_device,O_WRONLY); + else + audio_fd = open(audio_device,O_RDONLY); + if (audio_fd < 0) { + perror(audio_device); + return -EIO; + } + + /* non blocking mode */ + fcntl(audio_fd, F_SETFL, O_NONBLOCK); + +#if 0 + tmp=(NB_FRAGMENTS << 16) | FRAGMENT_BITS; + err=ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp); + if (err < 0) { + perror("SNDCTL_DSP_SETFRAGMENT"); + } +#endif + + tmp=AFMT_S16_LE; + err=ioctl(audio_fd,SNDCTL_DSP_SETFMT,&tmp); + if (err < 0) { + perror("SNDCTL_DSP_SETFMT"); + goto fail; + } + + tmp= (channels == 2); + err=ioctl(audio_fd,SNDCTL_DSP_STEREO,&tmp); + if (err < 0) { + perror("SNDCTL_DSP_STEREO"); + goto fail; + } + + tmp = freq; + err=ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp); + if (err < 0) { + perror("SNDCTL_DSP_SPEED"); + goto fail; + } + + s->rate = tmp; + s->fd = audio_fd; + + return 0; + fail: + close(audio_fd); + free(s); + return -EIO; +} + +static int audio_close(URLContext *h) +{ + AudioData *s = h->priv_data; + + close(s->fd); + free(s); + return 0; +} + +URLProtocol audio_protocol = { + "audio", + audio_open, + audio_read, + audio_write, + NULL, /* seek */ + audio_close, + audio_get_format, +}; |