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gstreamer/sys/oss/gstosssrc.c
Thomas Vander Stichele 3ecf433432 expand tabs 
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2005-12-06 19:44:58 +00:00

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* gstosssrc.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#include <sys/soundcard.h>
#include "gstosssrc.h"
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
static GstElementDetails gst_oss_src_details =
GST_ELEMENT_DETAILS ("Audio Source (OSS)",
"Source/Audio",
"Capture from a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
#define DEFAULT_DEVICE "/dev/dsp"
#define DEFAULT_DEVICE_NAME ""
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
};
GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc,
GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer);
static void gst_oss_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_oss_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_oss_src_dispose (GObject * object);
static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc);
static gboolean gst_oss_src_open (GstAudioSrc * asrc);
static gboolean gst_oss_src_close (GstAudioSrc * asrc);
static gboolean gst_oss_src_prepare (GstAudioSrc * asrc,
GstRingBufferSpec * spec);
static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc);
static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length);
static guint gst_oss_src_delay (GstAudioSrc * asrc);
static void gst_oss_src_reset (GstAudioSrc * asrc);
static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
static void
gst_oss_src_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_oss_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_oss_src_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&osssrc_src_factory));
}
static void
gst_oss_src_class_init (GstOssSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss_src_dispose);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss_src_get_property);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss_src_set_property);
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_reset);
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"OSS device (usually /dev/dspN)", DEFAULT_DEVICE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
G_PARAM_READABLE));
}
static void
gst_oss_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOssSrc *src;
src = GST_OSS_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
if (src->device)
g_free (src->device);
src->device = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_oss_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOssSrc *src;
src = GST_OSS_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, src->device);
break;
case PROP_DEVICE_NAME:
g_value_set_string (value, src->device_name);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_oss_src_init (GstOssSrc * osssrc, GstOssSrcClass * g_class)
{
GST_DEBUG ("initializing osssrc");
osssrc->fd = -1;
osssrc->device = g_strdup (DEFAULT_DEVICE);
osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME);
}
static GstCaps *
gst_oss_src_getcaps (GstBaseSrc * bsrc)
{
GstOssSrc *osssrc;
GstCaps *caps;
osssrc = GST_OSS_SRC (bsrc);
if (osssrc->fd == -1) {
caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD
(bsrc)));
} else {
caps = gst_oss_helper_probe_caps (osssrc->fd);
}
return caps;
}
static gint
ilog2 (gint x)
{
/* well... hacker's delight explains... */
x = x | (x >> 1);
x = x | (x >> 2);
x = x | (x >> 4);
x = x | (x >> 8);
x = x | (x >> 16);
x = x - ((x >> 1) & 0x55555555);
x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
x = (x + (x >> 4)) & 0x0f0f0f0f;
x = x + (x >> 8);
x = x + (x >> 16);
return (x & 0x0000003f) - 1;
}
#define SET_PARAM(_oss, _name, _val) \
G_STMT_START { \
int _tmp = _val; \
if (ioctl(_oss->fd, _name, &_tmp) == -1) { \
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, \
("Unable to set param "G_STRINGIFY (_name)": %s", \
g_strerror (errno)), \
(NULL)); \
return FALSE; \
} \
GST_DEBUG_OBJECT (_oss, G_STRINGIFY (_name)" %d", _tmp); \
} G_STMT_END
#define GET_PARAM(_oss, _name, _val) \
G_STMT_START { \
if (ioctl(oss->fd, _name, _val) == -1) { \
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, \
("Unable to get param "G_STRINGIFY (_name)": %s", \
g_strerror (errno)), \
(NULL)); \
return FALSE; \
} \
GST_DEBUG_OBJECT (_oss, G_STRINGIFY (_name)" %d", _val); \
} G_STMT_END
static gint
gst_oss_src_get_format (GstBufferFormat fmt)
{
gint result;
switch (fmt) {
case GST_MU_LAW:
result = AFMT_MU_LAW;
break;
case GST_A_LAW:
result = AFMT_A_LAW;
break;
case GST_IMA_ADPCM:
result = AFMT_IMA_ADPCM;
break;
case GST_U8:
result = AFMT_U8;
break;
case GST_S16_LE:
result = AFMT_S16_LE;
break;
case GST_S16_BE:
result = AFMT_S16_BE;
break;
case GST_S8:
result = AFMT_S8;
break;
case GST_U16_LE:
result = AFMT_U16_LE;
break;
case GST_U16_BE:
result = AFMT_U16_BE;
break;
case GST_MPEG:
result = AFMT_MPEG;
break;
default:
result = 0;
break;
}
return result;
}
static gboolean
gst_oss_src_open (GstAudioSrc * asrc)
{
GstOssSrc *oss;
int mode;
oss = GST_OSS_SRC (asrc);
mode = O_RDONLY;
mode |= O_NONBLOCK;
oss->fd = open (oss->device, mode, 0);
if (oss->fd == -1)
goto open_failed;
if (!oss->mixer) {
oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
if (oss->mixer) {
g_free (oss->device_name);
oss->device_name = g_strdup (oss->mixer->cardname);
}
}
return TRUE;
open_failed:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unable to open device %s for recording: %s",
oss->device, g_strerror (errno)), (NULL));
return FALSE;
}
}
static gboolean
gst_oss_src_close (GstAudioSrc * asrc)
{
GstOssSrc *oss;
oss = GST_OSS_SRC (asrc);
close (oss->fd);
if (oss->mixer) {
gst_ossmixer_free (oss->mixer);
oss->mixer = NULL;
}
return TRUE;
}
static gboolean
gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
GstOssSrc *oss;
struct audio_buf_info info;
int mode;
int fmt, tmp;
oss = GST_OSS_SRC (asrc);
mode = fcntl (oss->fd, F_GETFL);
mode &= ~O_NONBLOCK;
if (fcntl (oss->fd, F_SETFL, mode) == -1)
goto non_block;
fmt = gst_oss_src_get_format (spec->format);
if (fmt == 0)
goto wrong_format;
tmp = ilog2 (spec->segsize);
tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
GST_DEBUG ("set segsize: %d, segtotal: %d, value: %08x", spec->segsize,
spec->segtotal, tmp);
SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp);
SET_PARAM (oss, SNDCTL_DSP_RESET, 0);
SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt);
if (spec->channels == 2)
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1);
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels);
SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate);
GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info);
spec->segsize = info.fragsize;
spec->segtotal = info.fragstotal;
if (spec->width != 16 && spec->width != 8)
goto dodgy_width;
spec->bytes_per_sample = (spec->width / 8) * spec->channels;
oss->bytes_per_sample = (spec->width / 8) * spec->channels;
memset (spec->silence_sample, 0, spec->bytes_per_sample);
GST_DEBUG ("got segsize: %d, segtotal: %d, value: %08x", spec->segsize,
spec->segtotal, tmp);
return TRUE;
non_block:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unable to set device %s in non blocking mode: %s",
oss->device, g_strerror (errno)), (NULL));
return FALSE;
}
wrong_format:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unable to get format %d", spec->format), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unexpected width %d", spec->width), (NULL));
return FALSE;
}
}
static gboolean
gst_oss_src_unprepare (GstAudioSrc * asrc)
{
/* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
if (!gst_oss_src_close (asrc))
goto couldnt_close;
if (!gst_oss_src_open (asrc))
goto couldnt_reopen;
return TRUE;
couldnt_close:
{
GST_DEBUG ("Could not close the audio device");
return FALSE;
}
couldnt_reopen:
{
GST_DEBUG ("Could not reopen the audio device");
return FALSE;
}
}
static guint
gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length)
{
return read (GST_OSS_SRC (asrc)->fd, data, length);
}
static guint
gst_oss_src_delay (GstAudioSrc * asrc)
{
GstOssSrc *oss;
gint delay = 0;
gint ret;
oss = GST_OSS_SRC (asrc);
#ifdef SNDCTL_DSP_GETODELAY
ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
#else
ret = -1;
#endif
if (ret < 0) {
audio_buf_info info;
ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
}
return delay / oss->bytes_per_sample;
}
static void
gst_oss_src_reset (GstAudioSrc * asrc)
{
GstOssSrc *oss;
//gint ret;
oss = GST_OSS_SRC (asrc);
/* deadlocks on my machine... */
//ret = ioctl (oss->fd, SNDCTL_DSP_RESET, 0);
}
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