Initial revision

Originally committed as revision 5 to svn://svn.ffmpeg.org/ffmpeg/trunk

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,
+};