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gstreamer/sys/oss/gstosssrc.c
Andy Wingo afb7bcab51 some s/KHz/Hz/ fixes in osssrc consistency-with-the-rest-of-gst adjustments 
Original commit message from CVS:
* some s/KHz/Hz/ fixes in osssrc
* added dxr3 plugin from rehan khwaja <rehankhwaja@yahoo.com>, with some
consistency-with-the-rest-of-gst adjustments
2002-03-29 00:23:16 +00:00

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* 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.
*/
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#include <sys/ioctl.h>
#include <unistd.h>
#include <gstosssrc.h>
static GstElementDetails gst_osssrc_details = {
"Audio Source (OSS)",
"Source/Audio",
"Read from the sound card",
VERSION,
"Erik Walthinsen <omega@cse.ogi.edu>",
"(C) 1999",
};
/* OssSrc signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_DEVICE,
ARG_BYTESPERREAD,
ARG_CUROFFSET,
ARG_FORMAT,
ARG_CHANNELS,
ARG_FREQUENCY
};
GST_PADTEMPLATE_FACTORY (osssrc_src_factory,
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"osssrc_src",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_INT (G_BYTE_ORDER),
"signed", GST_PROPS_LIST (
GST_PROPS_BOOLEAN (TRUE),
GST_PROPS_BOOLEAN (FALSE)
),
"width", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16)
),
"depth", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16)
),
"rate", GST_PROPS_INT_RANGE (1000, 48000),
"channels", GST_PROPS_INT_RANGE (1, 2)
)
)
static void gst_osssrc_class_init (GstOssSrcClass *klass);
static void gst_osssrc_init (GstOssSrc *osssrc);
static void gst_osssrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
static void gst_osssrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
static GstElementStateReturn gst_osssrc_change_state (GstElement *element);
static void gst_osssrc_close_audio (GstOssSrc *src);
static gboolean gst_osssrc_open_audio (GstOssSrc *src);
static void gst_osssrc_sync_parms (GstOssSrc *osssrc);
static GstBuffer * gst_osssrc_get (GstPad *pad);
static GstElementClass *parent_class = NULL;
/*static guint gst_osssrc_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_osssrc_get_type (void)
{
static GType osssrc_type = 0;
if (!osssrc_type) {
static const GTypeInfo osssrc_info = {
sizeof(GstOssSrcClass),
NULL,
NULL,
(GClassInitFunc)gst_osssrc_class_init,
NULL,
NULL,
sizeof(GstOssSrc),
0,
(GInstanceInitFunc)gst_osssrc_init,
};
osssrc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstOssSrc", &osssrc_info, 0);
}
return osssrc_type;
}
static void
gst_osssrc_class_init (GstOssSrcClass *klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BYTESPERREAD,
g_param_spec_ulong("bytes_per_read","bytes_per_read","bytes_per_read",
0,G_MAXULONG,0,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_CUROFFSET,
g_param_spec_ulong("curoffset","curoffset","curoffset",
0,G_MAXULONG,0,G_PARAM_READABLE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FORMAT,
g_param_spec_int("format","format","format",
G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_CHANNELS,
g_param_spec_int("channels","channels","channels",
G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY,
g_param_spec_int("frequency","frequency","frequency",
G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_DEVICE,
g_param_spec_string("device","device","oss device (/dev/dspN usually)",
"default",G_PARAM_READWRITE));
gobject_class->set_property = gst_osssrc_set_property;
gobject_class->get_property = gst_osssrc_get_property;
gstelement_class->change_state = gst_osssrc_change_state;
}
static void
gst_osssrc_init (GstOssSrc *osssrc)
{
osssrc->srcpad = gst_pad_new_from_template (
GST_PADTEMPLATE_GET (osssrc_src_factory), "src");
gst_pad_set_get_function(osssrc->srcpad,gst_osssrc_get);
gst_element_add_pad (GST_ELEMENT (osssrc), osssrc->srcpad);
osssrc->device = g_strdup ("/dev/dsp");
osssrc->fd = -1;
/* adding some default values */
osssrc->format = AFMT_S16_LE;
osssrc->channels = 2;
osssrc->frequency = 44100;
osssrc->bytes_per_read = 4096;
osssrc->curoffset = 0;
osssrc->basetime = 0;
osssrc->samples_since_basetime = 0;
}
static GstBuffer *
gst_osssrc_get (GstPad *pad)
{
GstOssSrc *src;
GstBuffer *buf;
glong readbytes;
glong readsamples;
g_return_val_if_fail (pad != NULL, NULL);
src = GST_OSSSRC(gst_pad_get_parent (pad));
GST_DEBUG (GST_CAT_PLUGIN_INFO, "attempting to read something from soundcard");
buf = gst_buffer_new ();
g_return_val_if_fail (buf, NULL);
GST_BUFFER_DATA (buf) = (gpointer)g_malloc (src->bytes_per_read);
readbytes = read (src->fd,GST_BUFFER_DATA (buf),
src->bytes_per_read);
if (readbytes == 0) {
gst_element_set_eos (GST_ELEMENT (src));
return NULL;
}
if (!GST_PAD_CAPS (pad)) {
/* set caps on src pad */
if (!gst_pad_try_set_caps (src->srcpad,
GST_CAPS_NEW (
"oss_src",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0), /*FIXME */
"endianness", GST_PROPS_INT (G_BYTE_ORDER), /*FIXME */
"signed", GST_PROPS_BOOLEAN (TRUE), /*FIXME */
"width", GST_PROPS_INT (src->format),
"depth", GST_PROPS_INT (src->format),
"rate", GST_PROPS_INT (src->frequency),
"channels", GST_PROPS_INT (src->channels)
)))
{
gst_element_error (GST_ELEMENT (src), "could not set caps");
return NULL;
}
}
GST_BUFFER_SIZE (buf) = readbytes;
GST_BUFFER_OFFSET (buf) = src->curoffset;
GST_BUFFER_TIMESTAMP (buf) = src->basetime +
src->samples_since_basetime * 1000000LL / src->frequency;
src->curoffset += readbytes;
readsamples = readbytes / src->channels;
if (src->format == 16) readsamples /= 2;
src->samples_since_basetime += readsamples;
GST_DEBUG (GST_CAT_PLUGIN_INFO, "pushed buffer from soundcard of %ld bytes, timestamp %lld", readbytes, GST_BUFFER_TIMESTAMP (buf));
return buf;
}
static void
gst_osssrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
{
GstOssSrc *src;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_OSSSRC (object));
src = GST_OSSSRC (object);
switch (prop_id) {
case ARG_BYTESPERREAD:
src->bytes_per_read = g_value_get_ulong (value);
break;
case ARG_FORMAT:
src->format = g_value_get_int (value);
break;
case ARG_CHANNELS:
src->channels = g_value_get_int (value);
break;
case ARG_FREQUENCY:
/* Preserve the timestamps */
src->basetime = src->samples_since_basetime * 1000000LL / src->frequency;
src->samples_since_basetime = 0;
src->frequency = g_value_get_int (value);
break;
case ARG_CUROFFSET:
src->curoffset = g_value_get_ulong (value);
break;
case ARG_DEVICE:
g_free(src->device);
src->device = g_strdup (g_value_get_string (value));
break;
default:
break;
}
}
static void
gst_osssrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
{
GstOssSrc *src;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_OSSSRC (object));
src = GST_OSSSRC (object);
switch (prop_id) {
case ARG_BYTESPERREAD:
g_value_set_ulong (value, src->bytes_per_read);
break;
case ARG_FORMAT:
g_value_set_int (value, src->format);
break;
case ARG_CHANNELS:
g_value_set_int (value, src->channels);
break;
case ARG_FREQUENCY:
g_value_set_int (value, src->frequency);
break;
case ARG_CUROFFSET:
g_value_set_ulong (value, src->curoffset);
break;
case ARG_DEVICE:
g_value_set_string (value, src->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstElementStateReturn
gst_osssrc_change_state (GstElement *element)
{
/* GstOssSrc *src = GST_OSSSRC (element); */
g_return_val_if_fail (GST_IS_OSSSRC (element), FALSE);
GST_DEBUG (GST_CAT_PLUGIN_INFO, "osssrc: state change");
/* if going down into NULL state, close the file if it's open */
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
if (GST_FLAG_IS_SET (element, GST_OSSSRC_OPEN))
gst_osssrc_close_audio (GST_OSSSRC (element));
/* otherwise (READY or higher) we need to open the sound card */
} else {
GST_DEBUG (GST_CAT_PLUGIN_INFO, "DEBUG: osssrc: ready or higher");
if (!GST_FLAG_IS_SET (element, GST_OSSSRC_OPEN)) {
if (!gst_osssrc_open_audio (GST_OSSSRC (element)))
return GST_STATE_FAILURE;
else
{
GST_DEBUG (GST_CAT_PLUGIN_INFO, "osssrc: device opened successfully");
/* thomas: we can't set caps here because the element is
* not actually ready yet */
}
}
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static gboolean
gst_osssrc_open_audio (GstOssSrc *src)
{
g_return_val_if_fail (!GST_FLAG_IS_SET (src, GST_OSSSRC_OPEN), FALSE);
/* first try to open the sound card */
src->fd = open(src->device, O_RDONLY);
/* if we have it, set the default parameters and go have fun */
if (src->fd > 0) {
/* set card state */
gst_osssrc_sync_parms (src);
GST_DEBUG (GST_CAT_PLUGIN_INFO,"opened audio: %s",src->device);
GST_FLAG_SET (src, GST_OSSSRC_OPEN);
return TRUE;
}
return FALSE;
}
static void
gst_osssrc_close_audio (GstOssSrc *src)
{
g_return_if_fail (GST_FLAG_IS_SET (src, GST_OSSSRC_OPEN));
close(src->fd);
src->fd = -1;
GST_FLAG_UNSET (src, GST_OSSSRC_OPEN);
}
static void
gst_osssrc_sync_parms (GstOssSrc *osssrc)
{
audio_buf_info ispace;
gint frag;
/* remember : ioctl on samplerate returns the sample rate the card
* is actually set to ! Setting it to 44101 Hz could cause it to
* be set to 44101, for example
*/
guint frequency;
g_return_if_fail (osssrc != NULL);
g_return_if_fail (GST_IS_OSSSRC (osssrc));
g_return_if_fail (osssrc->fd > 0);
frequency = osssrc->frequency;
frag = 0x7fff0006;
ioctl(osssrc->fd, SNDCTL_DSP_SETFRAGMENT, &frag);
ioctl(osssrc->fd, SNDCTL_DSP_RESET, 0);
ioctl(osssrc->fd, SNDCTL_DSP_SETFMT, &osssrc->format);
ioctl(osssrc->fd, SNDCTL_DSP_CHANNELS, &osssrc->channels);
ioctl(osssrc->fd, SNDCTL_DSP_SPEED, &osssrc->frequency);
osssrc->frequency = frequency;
ioctl(osssrc->fd, SNDCTL_DSP_SPEED, &frequency);
ioctl(osssrc->fd, SNDCTL_DSP_GETISPACE, &ispace);
ioctl(osssrc->fd, SNDCTL_DSP_GETBLKSIZE, &frag);
g_print("setting sound card to %dHz %d bit %s (%d bytes buffer, %d fragment)\n",
osssrc->frequency, osssrc->format,
(osssrc->channels == 2) ? "stereo" : "mono", ispace.bytes, frag);
}
gboolean
gst_osssrc_factory_init (GstPlugin *plugin)
{
GstElementFactory *factory;
factory = gst_elementfactory_new ("osssrc", GST_TYPE_OSSSRC, &gst_osssrc_details);
g_return_val_if_fail (factory != NULL, FALSE);
gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (osssrc_src_factory));
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
return TRUE;
}
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