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sys/oss/: Port from THREADED+wim's fixes.

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Original commit message from CVS:
2005-07-08  Andy Wingo  <wingo@pobox.com>

* sys/oss/: Port from THREADED+wim's fixes.
This commit is contained in:
Andy Wingo 2005-07-08 11:19:19 +00:00
parent 266b874436
commit 6fc2023e8f
14 changed files with 700 additions and 857 deletions
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2
ChangeLog
View file

@ -1,5 +1,7 @@
2005-07-08 Andy Wingo <wingo@pobox.com>
* sys/oss/: Port from THREADED+wim's fixes.
* gst/avi/Makefile.am (libgstavi_la_CFLAGS): No gettext hacks, the
defines come from config.h.

14
sys/Makefile.am
View file

@ -4,11 +4,11 @@
# DXR3_DIR=
# endif
# if USE_OSS
# OSS_DIR=oss
# else
if USE_OSS
OSS_DIR=oss
else
OSS_DIR=
# endif
endif
# if USE_OSX_AUDIO
# OSX_AUDIO_DIR=osxaudio
@ -34,12 +34,6 @@ OSS_DIR=
# SUNAUDIO_DIR=
# endif
# if USE_GST_V4L
# V4L_DIR=v4l
# else
# V4L_DIR=
# endif
# if USE_GST_V4L2
# V4L2_DIR=v4l2
# else

30
sys/oss/Makefile.am
View file

@ -2,24 +2,28 @@ plugin_LTLIBRARIES = libgstossaudio.la
libgstossaudio_la_SOURCES = gstossaudio.c \
gstosselement.c \
gstosshelper.c \
gstossmixer.c \
gstosssink.c \
gstosssrc.c
gstosssink.c
libgstossaudio_la_CFLAGS = $(GST_CFLAGS)
libgstossaudio_la_LIBADD = $(top_builddir)/gst-libs/gst/libgstinterfaces-@GST_MAJORMINOR@.la
libgstossaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(GST_PLUGINS_BASE_LIBS)
# gstossdmabuffer.c
libgstossaudio_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
libgstossaudio_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
-lgstinterfaces-@GST_MAJORMINOR@ \
-lgstaudio-@GST_MAJORMINOR@
libgstossaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstosssink.h \
gstosssrc.h \
gstosselement.h\
gstossmixer.h \
gst-i18n-plugin.h \
gettext.h
gstosselement.h \
gstosshelper.h \
gstossdmabuffer.h \
gstossmixer.h
noinst_PROGRAMS = oss_probe
# noinst_PROGRAMS = #oss_probe
oss_probe_SOURCES = oss_probe.c
oss_probe_CFLAGS = $(GST_CFLAGS)
oss_probe_LDADD = $(GLIB_LIBS)
# oss_probe_SOURCES = oss_probe.c
# oss_probe_CFLAGS = $(GST_CFLAGS)
# oss_probe_LDADD = $(GLIB_LIBS)

9
sys/oss/gstossaudio.c
View file

@ -21,7 +21,7 @@
#include "config.h"
#endif
#include "gst-i18n-plugin.h"
#include "gst/gst-i18n-plugin.h"
#include "gstosselement.h"
#include "gstosssink.h"
@ -34,13 +34,10 @@ GST_DEBUG_CATEGORY (oss_debug);
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_library_load ("gstaudio"))
return FALSE;
if (!gst_element_register (plugin, "ossmixer", GST_RANK_PRIMARY,
if ( /*!gst_element_register (plugin, "ossmixer", GST_RANK_PRIMARY,
GST_TYPE_OSSELEMENT) ||
!gst_element_register (plugin, "osssrc", GST_RANK_PRIMARY,
GST_TYPE_OSSSRC) ||
GST_TYPE_OSSSRC) || */
!gst_element_register (plugin, "osssink", GST_RANK_PRIMARY,
GST_TYPE_OSSSINK)) {
return FALSE;

316
sys/oss/gstossdmabuffer.c Normal file
View file

@ -0,0 +1,316 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstossdmabuffer.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 <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <string.h>
#include <sys/types.h>
#include <sys/mman.h>
#include <sys/soundcard.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <pthread.h>
#include "gstossdmabuffer.h"
static void gst_ossdmabuffer_class_init (GstOssDMABufferClass * klass);
static void gst_ossdmabuffer_init (GstOssDMABuffer * ossdmabuffer);
static void gst_ossdmabuffer_dispose (GObject * object);
static void gst_ossdmabuffer_finalize (GObject * object);
static gboolean gst_ossdmabuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_ossdmabuffer_release (GstRingBuffer * buf);
static gboolean gst_ossdmabuffer_play (GstRingBuffer * buf);
static gboolean gst_ossdmabuffer_stop (GstRingBuffer * buf);
static GstRingBufferClass *parent_class = NULL;
GType
gst_ossdmabuffer_get_type (void)
{
static GType ossdmabuffer_type = 0;
if (!ossdmabuffer_type) {
static const GTypeInfo ossdmabuffer_info = {
sizeof (GstOssDMABufferClass),
NULL,
NULL,
(GClassInitFunc) gst_ossdmabuffer_class_init,
NULL,
NULL,
sizeof (GstOssDMABuffer),
0,
(GInstanceInitFunc) gst_ossdmabuffer_init,
NULL
};
ossdmabuffer_type =
g_type_register_static (GST_TYPE_RINGBUFFER, "GstOssDMABuffer",
&ossdmabuffer_info, 0);
}
return ossdmabuffer_type;
}
static void
gst_ossdmabuffer_class_init (GstOssDMABufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_RINGBUFFER);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_ossdmabuffer_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_ossdmabuffer_finalize);
gstringbuffer_class->acquire = gst_ossdmabuffer_acquire;
gstringbuffer_class->release = gst_ossdmabuffer_release;
gstringbuffer_class->play = gst_ossdmabuffer_play;
gstringbuffer_class->stop = gst_ossdmabuffer_stop;
}
static void
gst_ossdmabuffer_init (GstOssDMABuffer * ossdmabuffer)
{
ossdmabuffer->cond = g_cond_new ();
ossdmabuffer->element = g_object_new (GST_TYPE_OSSELEMENT, NULL);
}
static void
gst_ossdmabuffer_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_ossdmabuffer_finalize (GObject * object)
{
GstOssDMABuffer *obuf = (GstOssDMABuffer *) object;
g_cond_free (obuf->cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_ossdmabuffer_func (GstRingBuffer * buf)
{
fd_set writeset;
struct count_info count;
GstOssDMABuffer *obuf = (GstOssDMABuffer *) buf;
GST_LOCK (buf);
while (obuf->running) {
if (buf->state == GST_RINGBUFFER_STATE_PLAYING) {
int segsize;
GST_UNLOCK (buf);
segsize = buf->spec.segsize;
FD_ZERO (&writeset);
FD_SET (obuf->fd, &writeset);
select (obuf->fd + 1, NULL, &writeset, NULL, NULL);
if (ioctl (obuf->fd, SNDCTL_DSP_GETOPTR, &count) == -1) {
perror ("GETOPTR");
continue;
}
if (count.blocks > buf->spec.segtotal)
count.blocks = buf->spec.segtotal;
gst_ringbuffer_callback (buf, count.blocks);
GST_LOCK (buf);
} else {
GST_OSSDMABUFFER_SIGNAL (obuf);
GST_OSSDMABUFFER_WAIT (obuf);
}
}
GST_UNLOCK (buf);
}
static gboolean
gst_ossdmabuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
int tmp;
struct audio_buf_info info;
GstOssDMABuffer *obuf = (GstOssDMABuffer *) buf;;
caddr_t mmap_buf;
int mode;
gint size;
gboolean parsed;
parsed = gst_osselement_parse_caps (obuf->element, spec->caps);
if (!parsed)
return FALSE;
mode = O_RDWR;
mode |= O_NONBLOCK;
obuf->fd = open ("/dev/dsp", mode, 0);
if (obuf->fd == -1) {
perror ("OPEN");
return FALSE;
}
//obuf->frag = 0x00040008;
obuf->frag = 0xffff000a;
tmp = obuf->element->format;
if (ioctl (obuf->fd, SNDCTL_DSP_SETFMT, &tmp) == -1) {
perror ("SETFMT");
return FALSE;
}
tmp = obuf->element->channels;
if (ioctl (obuf->fd, SNDCTL_DSP_STEREO, &tmp) == -1) {
perror ("STEREO");
return FALSE;
}
tmp = obuf->element->channels;
if (ioctl (obuf->fd, SNDCTL_DSP_CHANNELS, &tmp) == -1) {
perror ("CHANNELS");
return FALSE;
}
tmp = obuf->element->rate;
if (ioctl (obuf->fd, SNDCTL_DSP_SPEED, &tmp) == -1) {
perror ("SPEED");
return FALSE;
}
if (ioctl (obuf->fd, SNDCTL_DSP_GETCAPS, &obuf->caps) == -1) {
perror ("/dev/dsp");
fprintf (stderr, "Sorry but your sound driver is too old\n");
return FALSE;
}
if (!(obuf->caps & DSP_CAP_TRIGGER) || !(obuf->caps & DSP_CAP_MMAP)) {
fprintf (stderr, "Sorry but your soundcard can't do this\n");
return FALSE;
}
if (ioctl (obuf->fd, SNDCTL_DSP_SETFRAGMENT, &obuf->frag) == -1) {
perror ("SETFRAGMENT");
return FALSE;
}
if (ioctl (obuf->fd, SNDCTL_DSP_GETOSPACE, &info) == -1) {
perror ("GETOSPACE");
return FALSE;
}
buf->spec.segsize = info.fragsize;
buf->spec.segtotal = info.fragstotal;
size = info.fragsize * info.fragstotal;
g_print ("segsize %d, segtotal %d\n", info.fragsize, info.fragstotal);
mmap_buf = (caddr_t) mmap (NULL, size, PROT_WRITE,
MAP_FILE | MAP_SHARED, obuf->fd, 0);
if ((caddr_t) mmap_buf == (caddr_t) - 1) {
perror ("mmap (write)");
return FALSE;
}
buf->data = gst_buffer_new ();
GST_BUFFER_DATA (buf->data) = mmap_buf;
GST_BUFFER_SIZE (buf->data) = size;
GST_BUFFER_FLAG_SET (buf->data, GST_BUFFER_DONTFREE);
tmp = 0;
if (ioctl (obuf->fd, SNDCTL_DSP_SETTRIGGER, &tmp) == -1) {
perror ("SETTRIGGER");
return FALSE;
}
GST_LOCK (obuf);
obuf->running = TRUE;
obuf->thread = g_thread_create ((GThreadFunc) gst_ossdmabuffer_func,
buf, TRUE, NULL);
GST_OSSDMABUFFER_WAIT (obuf);
GST_UNLOCK (obuf);
return TRUE;
}
static gboolean
gst_ossdmabuffer_release (GstRingBuffer * buf)
{
GstOssDMABuffer *obuf = (GstOssDMABuffer *) buf;;
gst_buffer_unref (buf->data);
GST_LOCK (obuf);
obuf->running = FALSE;
GST_OSSDMABUFFER_SIGNAL (obuf);
GST_UNLOCK (obuf);
g_thread_join (obuf->thread);
return TRUE;
}
static gboolean
gst_ossdmabuffer_play (GstRingBuffer * buf)
{
int tmp;
GstOssDMABuffer *obuf;
obuf = (GstOssDMABuffer *) buf;
tmp = PCM_ENABLE_OUTPUT;
if (ioctl (obuf->fd, SNDCTL_DSP_SETTRIGGER, &tmp) == -1) {
perror ("SETTRIGGER");
}
GST_OSSDMABUFFER_SIGNAL (obuf);
return TRUE;
}
static gboolean
gst_ossdmabuffer_stop (GstRingBuffer * buf)
{
int tmp;
GstOssDMABuffer *obuf;
obuf = (GstOssDMABuffer *) buf;
tmp = 0;
if (ioctl (obuf->fd, SNDCTL_DSP_SETTRIGGER, &tmp) == -1) {
perror ("SETTRIGGER");
}
GST_OSSDMABUFFER_WAIT (obuf);
buf->playseg = 0;
return TRUE;
}

77
sys/oss/gstossdmabuffer.h Normal file
View file

@ -0,0 +1,77 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstossdmabuffer.h:
*
* 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.
*/
#ifndef __GST_OSSDMABUFFER_H__
#define __GST_OSSDMABUFFER_H__
#include <gst/gst.h>
#include "gstosselement.h"
#include <gst/audio/gstringbuffer.h>
G_BEGIN_DECLS
#define GST_TYPE_OSSDMABUFFER (gst_ossdmabuffer_get_type())
#define GST_OSSDMABUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSDMABUFFER,GstOssDMABuffer))
#define GST_OSSDMABUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSDMABUFFER,GstOssDMABufferClass))
#define GST_IS_OSSDMABUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSDMABUFFER))
#define GST_IS_OSSDMABUFFER_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSDMABUFFER))
#define GST_OSSELEMENT_GET(obj) GST_OSSELEMENT (obj->element)
typedef enum {
GST_OSSDMABUFFER_OPEN = (1 << 0),
} GstOssDMABufferFlags;
typedef struct _GstOssDMABuffer GstOssDMABuffer;
typedef struct _GstOssDMABufferClass GstOssDMABufferClass;
#define GST_OSSDMABUFFER_THREAD(buf) (GST_OSSDMABUFFER(buf)->thread)
#define GST_OSSDMABUFFER_LOCK GST_LOCK
#define GST_OSSDMABUFFER_UNLOCK GST_UNLOCK
#define GST_OSSDMABUFFER_COND(buf) (GST_OSSDMABUFFER(buf)->cond)
#define GST_OSSDMABUFFER_SIGNAL(buf) (g_cond_signal (GST_OSSDMABUFFER_COND (buf)))
#define GST_OSSDMABUFFER_WAIT(buf) (g_cond_wait (GST_OSSDMABUFFER_COND (buf), GST_GET_LOCK (buf)))
struct _GstOssDMABuffer {
GstRingBuffer buffer;
GstOssElement *element;
int fd;
int caps;
int frag;
GThread *thread;
GCond *cond;
gboolean running;
};
struct _GstOssDMABufferClass {
GstRingBufferClass parent_class;
};
GType gst_ossdmabuffer_get_type(void);
G_END_DECLS
#endif /* __GST_OSSDMABUFFER_H__ */

160
sys/oss/gstosselement.c
View file

@ -1,9 +1,8 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wim.taymans@chello.be>
* 2004 Toni Willberg <toniw@iki.fi>
*
* gstosselement.c:
* gstosssink.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@ -25,7 +24,7 @@
#include "config.h"
#endif
#include "gst-i18n-plugin.h"
#include "gst/gst-i18n-plugin.h"
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
@ -34,28 +33,16 @@
#include <errno.h>
#include <string.h>
#ifdef HAVE_OSS_INCLUDE_IN_SYS
#include <sys/soundcard.h>
#else
#ifdef HAVE_OSS_INCLUDE_IN_ROOT
#include <soundcard.h>
#else
#include <machine/soundcard.h>
#endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include <gst/propertyprobe/propertyprobe.h>
#include <gst/interfaces/propertyprobe.h>
#include "gstosselement.h"
#include "gstossmixer.h"
enum
{
ARG_0,
ARG_ZERO,
ARG_DEVICE,
ARG_MIXERDEV,
ARG_DEVICE_NAME
@ -79,7 +66,6 @@ static void gst_osselement_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_osselement_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstElementStateReturn gst_osselement_change_state (GstElement * element);
static GstElementClass *parent_class = NULL;
@ -152,6 +138,9 @@ gst_osselement_class_init (GstOssElementClass * klass)
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gobject_class->set_property = gst_osselement_set_property;
gobject_class->get_property = gst_osselement_get_property;
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));
@ -163,11 +152,7 @@ gst_osselement_class_init (GstOssElementClass * klass)
g_param_spec_string ("device_name", "Device name", "Name of the device",
NULL, G_PARAM_READABLE));
gobject_class->set_property = gst_osselement_set_property;
gobject_class->get_property = gst_osselement_get_property;
gobject_class->finalize = gst_osselement_finalize;
gstelement_class->change_state = gst_osselement_change_state;
}
static const GList *
@ -245,7 +230,7 @@ gst_osselement_class_probe_devices (GstOssElementClass * klass, gboolean check)
static GList *device_combinations;
GList *padtempllist;
gint openmode = O_RDONLY;
gboolean is_mixer = FALSE;
gboolean mixer = FALSE;
/* Ok, so how do we open the device? We assume that we have (max.) one
* pad, and if this is a sinkpad, we're osssink (w). else, we're osssrc
@ -257,7 +242,7 @@ gst_osselement_class_probe_devices (GstOssElementClass * klass, gboolean check)
if (GST_PAD_TEMPLATE_DIRECTION (firstpadtempl) == GST_PAD_SINK) {
openmode = O_WRONLY;
}
is_mixer = TRUE;
mixer = TRUE;
}
if (!init && !check) {
@ -303,7 +288,7 @@ gst_osselement_class_probe_devices (GstOssElementClass * klass, gboolean check)
/* we just check the dsp. we assume the mixer always works.
* we don't need a mixer anyway (says OSS)... If we are a
* mixer element, we use the mixer anyway. */
if ((fd = open (is_mixer ? mixer :
if ((fd = open (mixer ? mixer :
dsp, openmode | O_NONBLOCK)) > 0 || errno == EBUSY) {
GstOssDeviceCombination *combi;
@ -314,7 +299,7 @@ gst_osselement_class_probe_devices (GstOssElementClass * klass, gboolean check)
combi = g_new0 (GstOssDeviceCombination, 1);
combi->dsp = dsp;
combi->mixer = mixer;
combi->dev = is_mixer ? mixer_dev : dsp_dev;
combi->dev = mixer ? mixer_dev : dsp_dev;
device_combinations = device_combination_append (device_combinations,
combi);
} else {
@ -525,18 +510,19 @@ gst_osselement_parse_caps (GstOssElement * oss, const GstCaps * caps)
{
gint bps, format;
GstStructure *structure;
gboolean res;
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "width", &oss->width);
gst_structure_get_int (structure, "depth", &oss->depth);
res = gst_structure_get_int (structure, "width", &oss->width);
res &= gst_structure_get_int (structure, "depth", &oss->depth);
if (oss->width != oss->depth)
if (!res || oss->width != oss->depth)
return FALSE;
gst_structure_get_int (structure, "law", &oss->law);
gst_structure_get_int (structure, "endianness", &oss->endianness);
gst_structure_get_boolean (structure, "signed", &oss->sign);
res = gst_structure_get_int (structure, "law", &oss->law);
res &= gst_structure_get_int (structure, "endianness", &oss->endianness);
res &= gst_structure_get_boolean (structure, "signed", &oss->sign);
if (!gst_ossformat_get (oss->law, oss->endianness, oss->sign,
oss->width, oss->depth, &format, &bps)) {
@ -605,8 +591,10 @@ gst_osselement_sync_parms (GstOssElement * oss)
/* gint fragscale, frag_ln; */
if (oss->fd == -1)
if (oss->fd == -1) {
GST_INFO ("osselement: no fd");
return FALSE;
}
if ((oss->fragment & 0xFFFF) == 0) {
frag = 0;
@ -634,7 +622,7 @@ gst_osselement_sync_parms (GstOssElement * oss)
ioctl (oss->fd, SNDCTL_DSP_GETBLKSIZE, &oss->fragment_size);
if (oss->mode == GST_OSSELEMENT_WRITE) {
if (oss->mode == 1) {
ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &space);
} else {
ioctl (oss->fd, SNDCTL_DSP_GETISPACE, &space);
@ -680,31 +668,19 @@ gst_osselement_sync_parms (GstOssElement * oss)
return TRUE;
}
static gboolean
gst_osselement_open_audio (GstOssElement * oss)
gboolean
gst_osselement_open_audio (GstOssElement * oss, GstOssOpenMode mode)
{
gint caps;
GstOssOpenMode mode = GST_OSSELEMENT_READ;
const GList *padlist;
g_return_val_if_fail (oss->fd == -1, FALSE);
GST_INFO ("osselement: attempting to open sound device");
/* Ok, so how do we open the device? We assume that we have (max.) one
* pad, and if this is a sinkpad, we're osssink (w). else, we're osssrc (r) */
padlist = gst_element_get_pad_list (GST_ELEMENT (oss));
if (padlist != NULL) {
GstPad *firstpad = padlist->data;
if (GST_PAD_IS_SINK (firstpad)) {
mode = GST_OSSELEMENT_WRITE;
}
} else {
if (mode == GST_OSS_MODE_MIXER)
goto do_mixer;
}
/* first try to open the sound card */
if (mode == GST_OSSELEMENT_WRITE) {
if (mode == 1) {
/* open non blocking first so that it returns immediatly with an error
* when we cannot get to the device */
oss->fd = open (oss->device, O_WRONLY | O_NONBLOCK);
@ -728,7 +704,7 @@ gst_osselement_open_audio (GstOssElement * oss)
break;
case EACCES:
case ETXTBSY:
if (mode == GST_OSSELEMENT_WRITE)
if (mode == 1)
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
(_("Could not access device \"%s\", check its permissions."),
oss->device), GST_ERROR_SYSTEM);
@ -745,8 +721,7 @@ gst_osselement_open_audio (GstOssElement * oss)
GST_ERROR_SYSTEM);
break;
default:
/* FIXME: strerror is not threadsafe */
if (mode == GST_OSSELEMENT_WRITE)
if (mode == 1)
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
(_("Could not open device \"%s\" for writing."), oss->device),
GST_ERROR_SYSTEM);
@ -823,17 +798,17 @@ gst_osselement_open_audio (GstOssElement * oss)
oss->caps = caps;
do_mixer:
gst_ossmixer_build_list (oss);
gst_ossmixer_build_list (NULL, NULL);
return TRUE;
}
static void
void
gst_osselement_close_audio (GstOssElement * oss)
{
gst_ossmixer_free_list (oss);
gst_ossmixer_free_list (NULL);
if (oss->probed_caps) {
gst_caps_free (oss->probed_caps);
gst_caps_unref (oss->probed_caps);
oss->probed_caps = NULL;
}
@ -913,7 +888,7 @@ gst_osselement_set_property (GObject * object,
case ARG_DEVICE:
/* disallow changing the device while it is opened
get_property("device") should return the right one */
if (gst_element_get_state (GST_ELEMENT (oss)) == GST_STATE_NULL) {
if (oss->fd == -1) {
g_free (oss->device);
oss->device = g_strdup (g_value_get_string (value));
@ -939,7 +914,7 @@ gst_osselement_set_property (GObject * object,
case ARG_MIXERDEV:
/* disallow changing the device while it is opened
get_property("mixerdev") should return the right one */
if (gst_element_get_state (GST_ELEMENT (oss)) == GST_STATE_NULL) {
if (oss->fd == -1) {
g_free (oss->mixer_dev);
oss->mixer_dev = g_strdup (g_value_get_string (value));
}
@ -971,34 +946,6 @@ gst_osselement_get_property (GObject * object,
}
}
static GstElementStateReturn
gst_osselement_change_state (GstElement * element)
{
GstOssElement *oss = GST_OSSELEMENT (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
if (!gst_osselement_open_audio (oss)) {
return GST_STATE_FAILURE;
}
GST_INFO ("osselement: opened sound device");
break;
case GST_STATE_READY_TO_NULL:
gst_osselement_close_audio (oss);
gst_osselement_reset (oss);
GST_INFO ("osselement: closed sound device");
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
/* rate probing code */
@ -1061,31 +1008,15 @@ gst_osselement_probe_caps (GstOssElement * oss)
GstStructure *structure;
unsigned int format_bit;
unsigned int format_mask;
GstCaps *caps;
gboolean mono_supported = FALSE;
gboolean stereo_supported = FALSE;
int n_channels;
if (oss->probed_caps != NULL)
return;
if (oss->fd == -1)
return;
/* FIXME test make sure we're not currently playing */
/* check if the device supports mono, stereo or both */
n_channels = 1;
ret = ioctl (oss->fd, SNDCTL_DSP_CHANNELS, &n_channels);
if (n_channels == 1)
mono_supported = TRUE;
n_channels = 2;
ret = ioctl (oss->fd, SNDCTL_DSP_CHANNELS, &n_channels);
if (n_channels == 2)
stereo_supported = TRUE;
/* FIXME test both mono and stereo */
format_mask = AFMT_U8 | AFMT_S16_LE | AFMT_S16_BE | AFMT_S8 |
AFMT_U16_LE | AFMT_U16_BE;
@ -1101,12 +1032,7 @@ gst_osselement_probe_caps (GstOssElement * oss)
probe = g_new0 (GstOssProbe, 1);
probe->fd = oss->fd;
probe->format = format_bit;
if (stereo_supported) {
probe->n_channels = 2;
} else {
probe->n_channels = 1;
}
ret = gst_osselement_rate_probe_check (probe);
if (probe->min == -1 || probe->max == -1) {
@ -1140,21 +1066,7 @@ gst_osselement_probe_caps (GstOssElement * oss)
g_free (probe);
structure = gst_osselement_get_format_structure (format_bit);
if (mono_supported && stereo_supported) {
gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2,
NULL);
} else if (mono_supported) {
gst_structure_set (structure, "channels", G_TYPE_INT, 1, NULL);
} else if (stereo_supported) {
gst_structure_set (structure, "channels", G_TYPE_INT, 2, NULL);
} else {
/* falling back to [1,2] because we don't know what breaks if we abort here */
GST_ERROR (_("Your OSS device doesn't support mono or stereo."));
gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2,
NULL);
}
gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_structure_set_value (structure, "rate", &rate_value);
g_value_unset (&rate_value);

20
sys/oss/gstosselement.h
View file

@ -26,9 +26,7 @@
#include <gst/gst.h>
#include <sys/types.h>
/* debugging category */
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
#include "gstosshelper.h"
G_BEGIN_DECLS
@ -48,16 +46,6 @@ G_BEGIN_DECLS
typedef struct _GstOssElement GstOssElement;
typedef struct _GstOssElementClass GstOssElementClass;
typedef enum {
GST_OSSELEMENT_READ,
GST_OSSELEMENT_WRITE,
} GstOssOpenMode;
typedef struct _GstOssDeviceCombination {
gchar *dsp, *mixer;
dev_t dev;
} GstOssDeviceCombination;
struct _GstOssElement
{
/* yes, we're a gstelement too */
@ -73,7 +61,6 @@ struct _GstOssElement
gint fragment;
guint64 fragment_time;
gint fragment_size;
GstOssOpenMode mode;
GstCaps *probed_caps;
/* stats bytes per *second* */
@ -90,6 +77,7 @@ struct _GstOssElement
gint depth;
gint channels;
gint rate;
gint mode;
/* mixer stuff */
GList *tracklist;
@ -115,8 +103,12 @@ gboolean gst_osselement_parse_caps (GstOssElement *oss,
gboolean gst_osselement_merge_fixed_caps (GstOssElement *oss,
GstCaps *caps);
gboolean gst_osselement_open_audio (GstOssElement *oss, GstOssOpenMode mode);
gboolean gst_osselement_sync_parms (GstOssElement *oss);
void gst_osselement_reset (GstOssElement *oss);
void gst_osselement_close_audio (GstOssElement *oss);
gboolean gst_osselement_convert (GstOssElement *oss,
GstFormat src_format,

58
sys/oss/gstossmixer.c
View file

@ -30,22 +30,9 @@
#include <string.h>
#include <errno.h>
#include <sys/ioctl.h>
#ifdef HAVE_OSS_INCLUDE_IN_SYS
#include <sys/soundcard.h>
#else
#ifdef HAVE_OSS_INCLUDE_IN_ROOT
#include <soundcard.h>
#else
#include <machine/soundcard.h>
#endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include <gst-i18n-plugin.h>
#include <gst/gst-i18n-plugin.h>
#include "gstossmixer.h"
@ -202,7 +189,7 @@ gst_ossmixer_track_init (GstOssMixerTrack * track)
}
GstMixerTrack *
gst_ossmixer_track_new (GstOssElement * oss,
gst_ossmixer_track_new (GstOssDevice * oss,
gint track_num, gint max_chans, gint flags)
{
GstOssMixerTrack *osstrack;
@ -258,14 +245,17 @@ gst_ossmixer_interface_init (GstMixerClass * klass)
static gboolean
gst_ossmixer_supported (GstImplementsInterface * iface, GType iface_type)
{
GstOssDevice *oss = g_object_get_data (G_OBJECT (iface), "oss-data");
g_return_val_if_fail (oss != NULL, FALSE);
g_assert (iface_type == GST_TYPE_MIXER);
return (GST_OSSELEMENT (iface)->mixer_fd != -1);
return (oss->mixer_fd != -1);
}
/* unused with G_DISABLE_* */
static G_GNUC_UNUSED gboolean
gst_ossmixer_contains_track (GstOssElement * oss, GstOssMixerTrack * osstrack)
gst_ossmixer_contains_track (GstOssDevice * oss, GstOssMixerTrack * osstrack)
{
const GList *item;
@ -279,7 +269,9 @@ gst_ossmixer_contains_track (GstOssElement * oss, GstOssMixerTrack * osstrack)
static const GList *
gst_ossmixer_list_tracks (GstMixer * mixer)
{
return (const GList *) GST_OSSELEMENT (mixer)->tracklist;
GstOssDevice *oss = g_object_get_data (G_OBJECT (mixer), "oss-data");
return (const GList *) oss->tracklist;
}
static void
@ -287,10 +279,11 @@ gst_ossmixer_get_volume (GstMixer * mixer,
GstMixerTrack * track, gint * volumes)
{
gint volume;
GstOssElement *oss = GST_OSSELEMENT (mixer);
GstOssDevice *oss = g_object_get_data (G_OBJECT (mixer), "oss-data");
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
/* assert that we're opened and that we're using a known item */
g_return_if_fail (oss != NULL);
g_return_if_fail (oss->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (oss, osstrack));
@ -319,10 +312,11 @@ gst_ossmixer_set_volume (GstMixer * mixer,
GstMixerTrack * track, gint * volumes)
{
gint volume;
GstOssElement *oss = GST_OSSELEMENT (mixer);
GstOssDevice *oss = g_object_get_data (G_OBJECT (mixer), "oss-data");
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
/* assert that we're opened and that we're using a known item */
g_return_if_fail (oss != NULL);
g_return_if_fail (oss->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (oss, osstrack));
@ -351,10 +345,11 @@ static void
gst_ossmixer_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
{
int volume;
GstOssElement *oss = GST_OSSELEMENT (mixer);
GstOssDevice *oss = g_object_get_data (G_OBJECT (mixer), "oss-data");
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
/* assert that we're opened and that we're using a known item */
g_return_if_fail (oss != NULL);
g_return_if_fail (oss->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (oss, osstrack));
@ -384,10 +379,11 @@ static void
gst_ossmixer_set_record (GstMixer * mixer,
GstMixerTrack * track, gboolean record)
{
GstOssElement *oss = GST_OSSELEMENT (mixer);
GstOssDevice *oss = g_object_get_data (G_OBJECT (mixer), "oss-data");
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
/* assert that we're opened and that we're using a known item */
g_return_if_fail (oss != NULL);
g_return_if_fail (oss->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (oss, osstrack));
@ -430,11 +426,9 @@ gst_ossmixer_set_record (GstMixer * mixer,
}
void
gst_ossmixer_build_list (GstOssElement * oss)
gst_ossmixer_build_list (GstOssDeviceCombination * c, GstOssDevice * oss)
{
gint i, devmask, master = -1;
const GList *pads = gst_element_get_pad_list (GST_ELEMENT (oss));
GstPadDirection dir = GST_PAD_UNKNOWN;
#ifdef SOUND_MIXER_INFO
struct mixer_info minfo;
@ -442,18 +436,14 @@ gst_ossmixer_build_list (GstOssElement * oss)
g_return_if_fail (oss->mixer_fd == -1);
oss->mixer_fd = open (oss->mixer_dev, O_RDWR);
oss->mixer_fd = open (c->mixer, O_RDWR);
if (oss->mixer_fd == -1) {
/* this is valid. OSS devices don't need to expose a mixer */
GST_DEBUG ("Failed to open mixer device %s, mixing disabled: %s",
oss->mixer_dev, strerror (errno));
c->mixer, strerror (errno));
return;
}
/* get direction */
if (pads && g_list_length ((GList *) pads) == 1)
dir = GST_PAD_DIRECTION (GST_PAD (pads->data));
/* get masks */
if (ioctl (oss->mixer_fd, SOUND_MIXER_READ_RECMASK, &oss->recmask) < 0 ||
ioctl (oss->mixer_fd, SOUND_MIXER_READ_RECSRC, &oss->recdevs) < 0 ||
@ -499,8 +489,8 @@ gst_ossmixer_build_list (GstOssElement * oss)
record = TRUE;
/* do we want this in our list? */
if ((dir == GST_PAD_SRC && input == FALSE) ||
(dir == GST_PAD_SINK && i != SOUND_MIXER_PCM))
if ((oss->mode == GST_OSS_MODE_READ && input == FALSE) ||
(oss->mode == GST_OSS_MODE_WRITE && i != SOUND_MIXER_PCM))
continue;
/* add track to list */
@ -515,7 +505,7 @@ gst_ossmixer_build_list (GstOssElement * oss)
}
void
gst_ossmixer_free_list (GstOssElement * oss)
gst_ossmixer_free_list (GstOssDevice * oss)
{
if (oss->mixer_fd == -1)
return;

10
sys/oss/gstossmixer.h
View file

@ -23,8 +23,9 @@
#define __GST_OSS_MIXER_H__
#include <gst/gst.h>
#include <gst/mixer/mixer.h>
#include "gstosselement.h"
#include <gst/interfaces/mixer.h>
#include "gstosshelper.h"
G_BEGIN_DECLS
@ -56,8 +57,9 @@ GType gst_ossmixer_track_get_type (void);
void gst_ossmixer_interface_init (GstMixerClass *klass);
void gst_oss_interface_init (GstImplementsInterfaceClass *klass);
void gst_ossmixer_build_list (GstOssElement *oss);
void gst_ossmixer_free_list (GstOssElement *oss);
void gst_ossmixer_build_list (GstOssDeviceCombination * c,
GstOssDevice *oss);
void gst_ossmixer_free_list (GstOssDevice *oss);
G_END_DECLS

675
sys/oss/gstosssink.c
View file

@ -1,6 +1,6 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wim.taymans@chello.be>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* gstosssink.c:
*
@ -24,23 +24,12 @@
#include "config.h"
#endif
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#ifdef HAVE_OSS_INCLUDE_IN_SYS
#include <sys/soundcard.h>
#else
#ifdef HAVE_OSS_INCLUDE_IN_ROOT
#include <soundcard.h>
#else
#include <machine/soundcard.h>
#endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include "gstosssink.h"
@ -57,59 +46,33 @@ static void gst_osssink_class_init (GstOssSinkClass * klass);
static void gst_osssink_init (GstOssSink * osssink);
static void gst_osssink_dispose (GObject * object);
static GstElementStateReturn gst_osssink_change_state (GstElement * element);
static void gst_osssink_set_clock (GstElement * element, GstClock * clock);
static GstClock *gst_osssink_get_clock (GstElement * element);
static GstClockTime gst_osssink_get_time (GstClock * clock, gpointer data);
static GstCaps *gst_osssink_getcaps (GstBaseSink * bsink);
static const GstFormat *gst_osssink_get_formats (GstPad * pad);
static gboolean gst_osssink_convert (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
static const GstQueryType *gst_osssink_get_query_types (GstPad * pad);
static gboolean gst_osssink_query (GstElement * element, GstQueryType type,
GstFormat * format, gint64 * value);
static gboolean gst_osssink_sink_query (GstPad * pad, GstQueryType type,
GstFormat * format, gint64 * value);
static GstCaps *gst_osssink_sink_fixate (GstPad * pad, const GstCaps * caps);
static GstCaps *gst_osssink_getcaps (GstPad * pad);
static GstPadLinkReturn gst_osssink_sinkconnect (GstPad * pad,
const GstCaps * caps);
static void gst_osssink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_osssink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_osssink_chain (GstPad * pad, GstData * _data);
static gboolean gst_osssink_open (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_osssink_close (GstAudioSink * asink);
static guint gst_osssink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_osssink_delay (GstAudioSink * asink);
static void gst_osssink_reset (GstAudioSink * asink);
/* OssSink signals and args */
enum
{
SIGNAL_HANDOFF,
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_MUTE,
ARG_FRAGMENT,
ARG_BUFFER_SIZE,
ARG_SYNC,
ARG_CHUNK_SIZE
/* FILL ME */
};
static GstStaticPadTemplate osssink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
"signed = (boolean) { TRUE, FALSE }, "
//"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
//"signed = (boolean) { TRUE, FALSE }, "
"endianness = (int) LITTLE_ENDIAN, "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) { 8, 16 }, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
@ -119,7 +82,8 @@ static GstStaticPadTemplate osssink_sink_factory =
);
static GstElementClass *parent_class = NULL;
static guint gst_osssink_signals[LAST_SIGNAL] = { 0 };
/* static guint gst_osssink_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_osssink_get_type (void)
@ -140,7 +104,7 @@ gst_osssink_get_type (void)
};
osssink_type =
g_type_register_static (GST_TYPE_OSSELEMENT, "GstOssSink",
g_type_register_static (GST_TYPE_AUDIOSINK, "GstOssSink",
&osssink_info, 0);
}
@ -150,13 +114,6 @@ gst_osssink_get_type (void)
static void
gst_osssink_dispose (GObject * object)
{
GstOssSink *osssink = (GstOssSink *) object;
if (osssink->provided_clock) {
gst_object_unparent (GST_OBJECT (osssink->provided_clock));
osssink->provided_clock = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
@ -166,6 +123,7 @@ gst_osssink_base_init (gpointer g_class)
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_osssink_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&osssink_sink_factory));
}
@ -174,515 +132,188 @@ gst_osssink_class_init (GstOssSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_OSSELEMENT);
parent_class = g_type_class_ref (GST_TYPE_BASEAUDIOSINK);
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MUTE,
g_param_spec_boolean ("mute", "Mute", "Mute the audio",
FALSE, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SYNC,
g_param_spec_boolean ("sync", "Sync",
"If syncing on timestamps should be enabled", TRUE,
G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FRAGMENT,
g_param_spec_int ("fragment", "Fragment",
"The fragment as 0xMMMMSSSS (MMMM = total fragments, 2^SSSS = fragment size)",
0, G_MAXINT, 6, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BUFFER_SIZE,
g_param_spec_uint ("buffer_size", "Buffer size",
"Size of buffers in osssink's bufferpool (bytes)", 0, G_MAXINT, 4096,
G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_CHUNK_SIZE,
g_param_spec_uint ("chunk_size", "Chunk size",
"Write data in chunk sized buffers", 0, G_MAXUINT, 4096,
G_PARAM_READWRITE));
gst_osssink_signals[SIGNAL_HANDOFF] =
g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstOssSinkClass, handoff), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
gobject_class->set_property = gst_osssink_set_property;
gobject_class->get_property = gst_osssink_get_property;
gobject_class->dispose = gst_osssink_dispose;
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_osssink_change_state);
gstelement_class->query = GST_DEBUG_FUNCPTR (gst_osssink_query);
gstelement_class->set_clock = gst_osssink_set_clock;
gstelement_class->get_clock = gst_osssink_get_clock;
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_osssink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_osssink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_osssink_close);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_osssink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_osssink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_osssink_reset);
}
static void
gst_osssink_init (GstOssSink * osssink)
{
osssink->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&osssink_sink_factory), "sink");
gst_element_add_pad (GST_ELEMENT (osssink), osssink->sinkpad);
gst_pad_set_link_function (osssink->sinkpad, gst_osssink_sinkconnect);
gst_pad_set_getcaps_function (osssink->sinkpad, gst_osssink_getcaps);
gst_pad_set_fixate_function (osssink->sinkpad, gst_osssink_sink_fixate);
gst_pad_set_convert_function (osssink->sinkpad, gst_osssink_convert);
gst_pad_set_query_function (osssink->sinkpad, gst_osssink_sink_query);
gst_pad_set_query_type_function (osssink->sinkpad,
gst_osssink_get_query_types);
gst_pad_set_formats_function (osssink->sinkpad, gst_osssink_get_formats);
gst_pad_set_chain_function (osssink->sinkpad, gst_osssink_chain);
GST_DEBUG ("initializing osssink");
osssink->bufsize = 4096;
osssink->chunk_size = 4096;
osssink->mute = FALSE;
osssink->sync = TRUE;
osssink->provided_clock =
gst_audio_clock_new ("ossclock", gst_osssink_get_time, osssink);
gst_object_set_parent (GST_OBJECT (osssink->provided_clock),
GST_OBJECT (osssink));
osssink->handled = 0;
GST_FLAG_SET (osssink, GST_ELEMENT_THREAD_SUGGESTED);
GST_FLAG_SET (osssink, GST_ELEMENT_EVENT_AWARE);
osssink->element = g_object_new (GST_TYPE_OSSELEMENT, NULL);
}
static GstCaps *
gst_osssink_sink_fixate (GstPad * pad, const GstCaps * caps)
gst_osssink_getcaps (GstBaseSink * bsink)
{
GstCaps *newcaps;
GstStructure *structure;
newcaps =
gst_caps_new_full (gst_structure_copy (gst_caps_get_structure (caps, 0)),
NULL);
structure = gst_caps_get_structure (newcaps, 0);
if (gst_caps_structure_fixate_field_nearest_int (structure, "rate", 44100)) {
return newcaps;
}
if (gst_caps_structure_fixate_field_nearest_int (structure, "depth", 16)) {
return newcaps;
}
if (gst_caps_structure_fixate_field_nearest_int (structure, "width", 16)) {
return newcaps;
}
if (gst_caps_structure_fixate_field_nearest_int (structure, "channels", 2)) {
return newcaps;
}
gst_caps_free (newcaps);
return NULL;
}
static GstCaps *
gst_osssink_getcaps (GstPad * pad)
{
GstOssSink *osssink = GST_OSSSINK (gst_pad_get_parent (pad));
GstOssSink *osssink;
GstOssElement *element;
GstCaps *caps;
gst_osselement_probe_caps (GST_OSSELEMENT (osssink));
osssink = GST_OSSSINK (bsink);
element = osssink->element;
if (GST_OSSELEMENT (osssink)->probed_caps == NULL) {
caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
gst_osselement_probe_caps (element);
if (element->probed_caps == NULL) {
caps =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASESINK_PAD
(bsink)));
} else {
caps = gst_caps_copy (GST_OSSELEMENT (osssink)->probed_caps);
caps = gst_caps_ref (element->probed_caps);
}
return caps;
}
static GstPadLinkReturn
gst_osssink_sinkconnect (GstPad * pad, const GstCaps * caps)
static gint
ilog2 (gint x)
{
GstOssSink *osssink = GST_OSSSINK (gst_pad_get_parent (pad));
if (!gst_osselement_parse_caps (GST_OSSELEMENT (osssink), caps))
return GST_PAD_LINK_REFUSED;
if (!gst_osselement_sync_parms (GST_OSSELEMENT (osssink))) {
return GST_PAD_LINK_REFUSED;
}
return GST_PAD_LINK_OK;
/* 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;
}
static inline gint
gst_osssink_get_delay (GstOssSink * osssink)
#define SET_PARAM(_oss, _name, _val) \
G_STMT_START { \
int _tmp = _val; \
if (ioctl(_oss->fd, _name, &_tmp) == -1) { \
perror(G_STRINGIFY (_name)); \
return FALSE; \
} \
GST_DEBUG(G_STRINGIFY (name) " %d", _tmp); \
} G_STMT_END
#define GET_PARAM(oss, name, val) \
G_STMT_START { \
if (ioctl(oss->fd, name, val) == -1) { \
perror(G_STRINGIFY (name)); \
return FALSE; \
} \
} G_STMT_END
static gboolean
gst_osssink_open (GstAudioSink * asink, GstRingBufferSpec * spec)
{
struct audio_buf_info info;
int mode;
GstOssSink *oss;
int tmp;
oss = GST_OSSSINK (asink);
mode = O_WRONLY;
mode |= O_NONBLOCK;
oss->fd = open ("/dev/dsp", mode, 0);
if (oss->fd == -1) {
perror ("/dev/dsp");
return FALSE;
}
mode = fcntl (oss->fd, F_GETFL);
mode &= ~O_NONBLOCK;
fcntl (oss->fd, F_SETFL, mode);
SET_PARAM (oss, SNDCTL_DSP_SETFMT, AFMT_S16_LE);
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1);
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, 2);
SET_PARAM (oss, SNDCTL_DSP_SPEED, 44100);
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);
GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info);
spec->segsize = info.fragsize;
spec->segtotal = info.fragstotal;
spec->bytes_per_sample = 4;
oss->bytes_per_sample = 4;
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;
}
static gboolean
gst_osssink_close (GstAudioSink * asink)
{
close (GST_OSSSINK (asink)->fd);
return TRUE;
}
static guint
gst_osssink_write (GstAudioSink * asink, gpointer data, guint length)
{
return write (GST_OSSSINK (asink)->fd, data, length);
}
static guint
gst_osssink_delay (GstAudioSink * asink)
{
GstOssSink *oss;
gint delay = 0;
gint ret;
if (GST_OSSELEMENT (osssink)->fd == -1)
return 0;
oss = GST_OSSSINK (asink);
#ifdef SNDCTL_DSP_GETODELAY
ret = ioctl (GST_OSSELEMENT (osssink)->fd, SNDCTL_DSP_GETODELAY, &delay);
ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
#else
ret = -1;
#endif
if (ret < 0) {
audio_buf_info info;
if (ioctl (GST_OSSELEMENT (osssink)->fd, SNDCTL_DSP_GETOSPACE, &info) < 0) {
delay = 0;
} else {
delay = (info.fragstotal * info.fragsize) - info.bytes;
ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
}
}
return delay;
}
static GstClockTime
gst_osssink_get_time (GstClock * clock, gpointer data)
{
GstOssSink *osssink = GST_OSSSINK (data);
gint delay;
GstClockTime res;
if (!GST_OSSELEMENT (osssink)->bps)
return 0;
delay = gst_osssink_get_delay (osssink);
/* sometimes delay is bigger than the number of bytes sent to the device,
* which screws up this calculation, we assume that everything is still
* in the device then
* thomas: with proper handling of the return value, this doesn't seem to
* happen anymore, so remove the second code path after april 2004 */
if (delay > (gint64) osssink->handled) {
/*g_warning ("Delay %d > osssink->handled %" G_GUINT64_FORMAT
", setting to osssink->handled",
delay, osssink->handled); */
delay = osssink->handled;
}
res =
((gint64) osssink->handled -
delay) * GST_SECOND / GST_OSSELEMENT (osssink)->bps;
if (res < 0)
res = 0;
return res;
}
static GstClock *
gst_osssink_get_clock (GstElement * element)
{
GstOssSink *osssink;
osssink = GST_OSSSINK (element);
return GST_CLOCK (osssink->provided_clock);
return delay / oss->bytes_per_sample;
}
static void
gst_osssink_set_clock (GstElement * element, GstClock * clock)
gst_osssink_reset (GstAudioSink * asink)
{
GstOssSink *osssink;
GstOssSink *oss;
osssink = GST_OSSSINK (element);
//gint ret;
osssink->clock = clock;
}
static void
gst_osssink_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
GstOssSink *osssink;
GstClockTimeDiff buftime, soundtime, elementtime;
guchar *data;
guint to_write;
gint delay;
/* this has to be an audio buffer */
osssink = GST_OSSSINK (gst_pad_get_parent (pad));
if (GST_IS_EVENT (buf)) {
GstEvent *event = GST_EVENT (buf);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
ioctl (GST_OSSELEMENT (osssink)->fd, SNDCTL_DSP_SYNC, 0);
gst_audio_clock_set_active (GST_AUDIO_CLOCK (osssink->provided_clock),
FALSE);
gst_pad_event_default (pad, event);
return;
default:
gst_pad_event_default (pad, event);
return;
}
g_assert_not_reached ();
}
if (!GST_OSSELEMENT (osssink)->bps) {
gst_buffer_unref (buf);
GST_ELEMENT_ERROR (osssink, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before chain function"));
return;
}
data = GST_BUFFER_DATA (buf);
to_write = GST_BUFFER_SIZE (buf);
/* sync audio with buffers timestamp. elementtime is the *current* time.
* soundtime is the time if the soundcard has processed all queued data. */
elementtime = gst_element_get_time (GST_ELEMENT (osssink));
delay = gst_osssink_get_delay (osssink);
if (delay < 0)
delay = 0;
soundtime = elementtime + delay * GST_SECOND / GST_OSSELEMENT (osssink)->bps;
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
buftime = GST_BUFFER_TIMESTAMP (buf);
} else {
buftime = soundtime;
}
GST_LOG_OBJECT (osssink,
"time: real %" GST_TIME_FORMAT ", buffer: %" GST_TIME_FORMAT,
GST_TIME_ARGS (soundtime), GST_TIME_ARGS (buftime));
if (MAX (buftime, soundtime) - MIN (buftime, soundtime) > (GST_SECOND / 10)) {
/* we need to adjust to the buffers here */
GST_INFO_OBJECT (osssink,
"need sync: real %" GST_TIME_FORMAT ", buffer: %" GST_TIME_FORMAT,
GST_TIME_ARGS (soundtime), GST_TIME_ARGS (buftime));
if (soundtime > buftime) {
/* do *not* throw frames out. It's useless. The next frame will come in
* too late. And the next one. And so on. We don't want to lose sound.
* This is a placeholder for what - some day - should become QoS, i.e.
* sending events upstream to drop buffers. */
} else {
guint64 to_handle =
(((buftime -
soundtime) * GST_OSSELEMENT (osssink)->bps / GST_SECOND) /
((GST_OSSELEMENT (osssink)->width / 8) *
GST_OSSELEMENT (osssink)->channels)) *
(GST_OSSELEMENT (osssink)->width / 8) *
GST_OSSELEMENT (osssink)->channels;
guint8 *sbuf = g_new (guint8, to_handle);
memset (sbuf, (GST_OSSELEMENT (osssink)->width == 8) ? 0 : 128,
to_handle);
while (to_handle > 0) {
gint done = write (GST_OSSELEMENT (osssink)->fd, sbuf,
MIN (to_handle, osssink->chunk_size));
if (done == -1 && errno != EINTR) {
break;
} else {
to_handle -= done;
osssink->handled += done;
}
}
g_free (sbuf);
}
}
if (GST_OSSELEMENT (osssink)->fd >= 0 && to_write > 0) {
if (!osssink->mute) {
while (to_write > 0) {
gint done = write (GST_OSSELEMENT (osssink)->fd, data,
MIN (to_write, osssink->chunk_size));
if (done == -1) {
if (errno != EINTR)
break;
} else {
to_write -= done;
data += done;
osssink->handled += done;
}
}
} else {
g_warning ("muting osssinks unimplemented wrt clocks!");
}
}
gst_audio_clock_update_time ((GstAudioClock *) osssink->provided_clock,
gst_osssink_get_time (osssink->provided_clock, osssink));
gst_buffer_unref (buf);
}
static const GstFormat *
gst_osssink_get_formats (GstPad * pad)
{
static const GstFormat formats[] = {
GST_FORMAT_TIME,
GST_FORMAT_DEFAULT,
GST_FORMAT_BYTES,
0
};
return formats;
}
static gboolean
gst_osssink_convert (GstPad * pad, GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
GstOssSink *osssink;
osssink = GST_OSSSINK (gst_pad_get_parent (pad));
return gst_osselement_convert (GST_OSSELEMENT (osssink),
src_format, src_value, dest_format, dest_value);
}
static const GstQueryType *
gst_osssink_get_query_types (GstPad * pad)
{
static const GstQueryType query_types[] = {
GST_QUERY_LATENCY,
GST_QUERY_POSITION,
0,
};
return query_types;
}
static gboolean
gst_osssink_sink_query (GstPad * pad, GstQueryType type, GstFormat * format,
gint64 * value)
{
gboolean res = TRUE;
GstOssSink *osssink;
osssink = GST_OSSSINK (gst_pad_get_parent (pad));
switch (type) {
case GST_QUERY_LATENCY:
if (!gst_osssink_convert (pad,
GST_FORMAT_BYTES, gst_osssink_get_delay (osssink),
format, value)) {
res = FALSE;
}
break;
case GST_QUERY_POSITION:
if (!gst_osssink_convert (pad,
GST_FORMAT_TIME, gst_element_get_time (GST_ELEMENT (osssink)),
format, value)) {
res = FALSE;
}
break;
default:
res =
gst_pad_query (gst_pad_get_peer (osssink->sinkpad), type, format,
value);
break;
}
return res;
}
static gboolean
gst_osssink_query (GstElement * element, GstQueryType type, GstFormat * format,
gint64 * value)
{
GstOssSink *osssink = GST_OSSSINK (element);
return gst_osssink_sink_query (osssink->sinkpad, type, format, value);
}
static void
gst_osssink_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstOssSink *osssink;
osssink = GST_OSSSINK (object);
switch (prop_id) {
case ARG_MUTE:
osssink->mute = g_value_get_boolean (value);
g_object_notify (G_OBJECT (osssink), "mute");
break;
case ARG_FRAGMENT:
GST_OSSELEMENT (osssink)->fragment = g_value_get_int (value);
gst_osselement_sync_parms (GST_OSSELEMENT (osssink));
break;
case ARG_BUFFER_SIZE:
osssink->bufsize = g_value_get_uint (value);
g_object_notify (object, "buffer_size");
break;
case ARG_SYNC:
osssink->sync = g_value_get_boolean (value);
g_object_notify (G_OBJECT (osssink), "sync");
break;
case ARG_CHUNK_SIZE:
osssink->chunk_size = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_osssink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstOssSink *osssink;
osssink = GST_OSSSINK (object);
switch (prop_id) {
case ARG_MUTE:
g_value_set_boolean (value, osssink->mute);
break;
case ARG_FRAGMENT:
g_value_set_int (value, GST_OSSELEMENT (osssink)->fragment);
break;
case ARG_BUFFER_SIZE:
g_value_set_uint (value, osssink->bufsize);
break;
case ARG_SYNC:
g_value_set_boolean (value, osssink->sync);
break;
case ARG_CHUNK_SIZE:
g_value_set_uint (value, osssink->chunk_size);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstElementStateReturn
gst_osssink_change_state (GstElement * element)
{
GstOssSink *osssink;
osssink = GST_OSSSINK (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_READY_TO_PAUSED:
break;
case GST_STATE_PAUSED_TO_PLAYING:
gst_audio_clock_set_active (GST_AUDIO_CLOCK (osssink->provided_clock),
TRUE);
break;
case GST_STATE_PLAYING_TO_PAUSED:
if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
ioctl (GST_OSSELEMENT (osssink)->fd, SNDCTL_DSP_RESET, 0);
gst_audio_clock_set_active (GST_AUDIO_CLOCK (osssink->provided_clock),
FALSE);
break;
case GST_STATE_PAUSED_TO_READY:
if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
ioctl (GST_OSSELEMENT (osssink)->fd, SNDCTL_DSP_RESET, 0);
gst_osselement_reset (GST_OSSELEMENT (osssink));
osssink->handled = 0;
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
oss = GST_OSSSINK (asink);
/* deadlocks on my machine... */
//ret = ioctl (oss->fd, SNDCTL_DSP_RESET, 0);
}

42
sys/oss/gstosssink.h
View file

@ -26,52 +26,32 @@
#include <gst/gst.h>
#include <gst/audio/gstaudiosink.h>
#include "gstosselement.h"
#include <gst/audio/audioclock.h>
G_BEGIN_DECLS
#define GST_TYPE_OSSSINK \
(gst_osssink_get_type())
#define GST_OSSSINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSSINK,GstOssSink))
#define GST_OSSSINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSSINK,GstOssSinkClass))
#define GST_IS_OSSSINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSSINK))
#define GST_IS_OSSSINK_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSSINK))
typedef enum {
GST_OSSSINK_OPEN = GST_ELEMENT_FLAG_LAST,
GST_OSSSINK_FLAG_LAST = GST_ELEMENT_FLAG_LAST+2,
} GstOssSinkFlags;
#define GST_TYPE_OSSSINK (gst_osssink_get_type())
#define GST_OSSSINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSSINK,GstOssSink))
#define GST_OSSSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSSINK,GstOssSinkClass))
#define GST_IS_OSSSINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSSINK))
#define GST_IS_OSSSINK_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSSINK))
typedef struct _GstOssSink GstOssSink;
typedef struct _GstOssSinkClass GstOssSinkClass;
struct _GstOssSink {
GstOssElement element;
GstAudioSink sink;
GstPad *sinkpad;
GstOssElement *element;
GstClock *provided_clock;
GstClock *clock;
gboolean sync;
guint64 handled;
gboolean mute;
guint bufsize;
guint chunk_size;
gint fd;
gint bytes_per_sample;
};
struct _GstOssSinkClass {
GstOssElementClass parent_class;
/* signals */
void (*handoff) (GstElement *element,GstPad *pad);
GstAudioSinkClass parent_class;
};
GType gst_osssink_get_type(void);

72
sys/oss/gstosssrc.c
View file

@ -92,8 +92,7 @@ static void gst_osssrc_class_init (GstOssSrcClass * klass);
static void gst_osssrc_init (GstOssSrc * osssrc);
static void gst_osssrc_dispose (GObject * object);
static GstPadLinkReturn gst_osssrc_srcconnect (GstPad * pad,
const GstCaps * caps);
static GstPadLinkReturn gst_osssrc_src_link (GstPad * pad, GstPad * peer);
static GstCaps *gst_osssrc_getcaps (GstPad * pad);
static const GstFormat *gst_osssrc_get_formats (GstPad * pad);
static gboolean gst_osssrc_convert (GstPad * pad,
@ -117,7 +116,7 @@ static const GstQueryType *gst_osssrc_get_query_types (GstPad * pad);
static gboolean gst_osssrc_src_query (GstPad * pad, GstQueryType type,
GstFormat * format, gint64 * value);
static GstData *gst_osssrc_get (GstPad * pad);
static void gst_osssrc_loop (GstPad * pad);
static GstElementClass *parent_class = NULL;
@ -168,6 +167,9 @@ gst_osssrc_class_init (GstOssSrcClass * klass)
parent_class = g_type_class_ref (GST_TYPE_OSSELEMENT);
gobject_class->set_property = gst_osssrc_set_property;
gobject_class->get_property = gst_osssrc_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BUFFERSIZE,
g_param_spec_ulong ("buffersize", "Buffer Size",
"The size of the buffers with samples", 0, G_MAXULONG, 0,
@ -177,8 +179,6 @@ gst_osssrc_class_init (GstOssSrcClass * klass)
"The fragment as 0xMMMMSSSS (MMMM = total fragments, 2^SSSS = fragment size)",
0, G_MAXINT, 6, G_PARAM_READWRITE));
gobject_class->set_property = gst_osssrc_set_property;
gobject_class->get_property = gst_osssrc_get_property;
gobject_class->dispose = gst_osssrc_dispose;
gstelement_class->change_state = gst_osssrc_change_state;
@ -194,9 +194,9 @@ gst_osssrc_init (GstOssSrc * osssrc)
osssrc->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&osssrc_src_factory), "src");
gst_pad_set_get_function (osssrc->srcpad, gst_osssrc_get);
gst_pad_set_loop_function (osssrc->srcpad, gst_osssrc_loop);
gst_pad_set_getcaps_function (osssrc->srcpad, gst_osssrc_getcaps);
gst_pad_set_link_function (osssrc->srcpad, gst_osssrc_srcconnect);
gst_pad_set_link_function (osssrc->srcpad, gst_osssrc_src_link);
gst_pad_set_convert_function (osssrc->srcpad, gst_osssrc_convert);
gst_pad_set_formats_function (osssrc->srcpad, gst_osssrc_get_formats);
gst_pad_set_event_function (osssrc->srcpad, gst_osssrc_src_event);
@ -236,7 +236,7 @@ gst_osssrc_getcaps (GstPad * pad)
GstOssSrc *src;
GstCaps *caps;
src = GST_OSSSRC (gst_pad_get_parent (pad));
src = GST_OSSSRC (GST_PAD_PARENT (pad));
gst_osselement_probe_caps (GST_OSSELEMENT (src));
@ -250,19 +250,9 @@ gst_osssrc_getcaps (GstPad * pad)
}
static GstPadLinkReturn
gst_osssrc_srcconnect (GstPad * pad, const GstCaps * caps)
gst_osssrc_src_link (GstPad * pad, GstPad * peer)
{
GstOssSrc *src;
src = GST_OSSSRC (gst_pad_get_parent (pad));
if (!gst_osselement_parse_caps (GST_OSSELEMENT (src), caps))
return GST_PAD_LINK_REFUSED;
if (!gst_osselement_sync_parms (GST_OSSELEMENT (src)))
return GST_PAD_LINK_REFUSED;
return GST_PAD_LINK_OK;
return GST_RPAD_LINKFUNC (peer) (peer, pad);
}
static gboolean
@ -271,9 +261,10 @@ gst_osssrc_negotiate (GstPad * pad)
GstOssSrc *src;
GstCaps *allowed;
src = GST_OSSSRC (gst_pad_get_parent (pad));
src = GST_OSSSRC (GST_PAD_PARENT (pad));
allowed = gst_pad_get_allowed_caps (pad);
//allowed = gst_pad_get_allowed_caps (pad);
allowed = NULL;
if (!gst_osselement_merge_fixed_caps (GST_OSSELEMENT (src), allowed))
return FALSE;
@ -282,17 +273,15 @@ gst_osssrc_negotiate (GstPad * pad)
return FALSE;
/* set caps on src pad */
if (gst_pad_try_set_caps (src->srcpad,
GST_PAD_CAPS (src->srcpad) =
gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, GST_OSSELEMENT (src)->endianness,
"signed", G_TYPE_BOOLEAN, GST_OSSELEMENT (src)->sign,
"width", G_TYPE_INT, GST_OSSELEMENT (src)->width,
"depth", G_TYPE_INT, GST_OSSELEMENT (src)->depth,
"rate", G_TYPE_INT, GST_OSSELEMENT (src)->rate,
"channels", G_TYPE_INT, GST_OSSELEMENT (src)->channels,
NULL)) <= 0) {
return FALSE;
}
"channels", G_TYPE_INT, GST_OSSELEMENT (src)->channels, NULL);
return TRUE;
}
@ -332,21 +321,22 @@ gst_osssrc_set_clock (GstElement * element, GstClock * clock)
osssrc->clock = clock;
}
static GstData *
gst_osssrc_get (GstPad * pad)
static void
gst_osssrc_loop (GstPad * pad)
{
GstOssSrc *src;
GstBuffer *buf;
glong readbytes;
glong readsamples;
src = GST_OSSSRC (gst_pad_get_parent (pad));
src = GST_OSSSRC (GST_PAD_PARENT (pad));
GST_DEBUG ("attempting to read something from the soundcard");
if (src->need_eos) {
src->need_eos = FALSE;
return GST_DATA (gst_event_new (GST_EVENT_EOS));
gst_pad_push_event (pad, gst_event_new (GST_EVENT_EOS));
return;
}
buf = gst_buffer_new_and_alloc (src->buffersize);
@ -356,14 +346,14 @@ gst_osssrc_get (GstPad * pad)
if (!gst_osssrc_negotiate (pad)) {
gst_buffer_unref (buf);
GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL), (NULL));
return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
return;
}
}
if (GST_OSSELEMENT (src)->bps == 0) {
gst_buffer_unref (buf);
GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before chain function"));
return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
return;
}
readbytes = read (GST_OSSELEMENT (src)->fd, GST_BUFFER_DATA (buf),
@ -371,13 +361,13 @@ gst_osssrc_get (GstPad * pad)
if (readbytes < 0) {
gst_buffer_unref (buf);
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
return;
}
if (readbytes == 0) {
gst_buffer_unref (buf);
gst_element_set_eos (GST_ELEMENT (src));
return GST_DATA (gst_event_new (GST_EVENT_INTERRUPT));
gst_pad_push_event (pad, gst_event_new (GST_EVENT_EOS));
return;
}
readsamples = readbytes * GST_OSSELEMENT (src)->rate /
@ -412,7 +402,9 @@ gst_osssrc_get (GstPad * pad)
GST_DEBUG ("pushed buffer from soundcard of %ld bytes, timestamp %"
G_GINT64_FORMAT, readbytes, GST_BUFFER_TIMESTAMP (buf));
return GST_DATA (buf);
gst_pad_push (pad, buf);
return;
}
static void
@ -509,7 +501,7 @@ gst_osssrc_convert (GstPad * pad, GstFormat src_format, gint64 src_value,
{
GstOssSrc *osssrc;
osssrc = GST_OSSSRC (gst_pad_get_parent (pad));
osssrc = GST_OSSSRC (GST_PAD_PARENT (pad));
return gst_osselement_convert (GST_OSSELEMENT (osssrc), src_format, src_value,
dest_format, dest_value);
@ -533,7 +525,7 @@ gst_osssrc_src_event (GstPad * pad, GstEvent * event)
GstOssSrc *osssrc;
gboolean retval = FALSE;
osssrc = GST_OSSSRC (gst_pad_get_parent (pad));
osssrc = GST_OSSSRC (GST_PAD_PARENT (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
@ -589,7 +581,7 @@ gst_osssrc_src_query (GstPad * pad, GstQueryType type, GstFormat * format,
gboolean res = FALSE;
GstOssSrc *osssrc;
osssrc = GST_OSSSRC (gst_pad_get_parent (pad));
osssrc = GST_OSSSRC (GST_PAD_PARENT (pad));
switch (type) {
case GST_QUERY_POSITION:

48
sys/oss/oss_probe.c
View file

@ -1,24 +1,3 @@
/* GStreamer
* Copyright (C) 2004 David Schleef
* 2004 Toni Willberg <toniw@iki.fi>
*
* oss_probe.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"
@ -73,12 +52,9 @@ int
main (int argc, char *argv[])
{
int fd;
int i, ret;
int i;
Probe *probe;
gboolean mono_supported = FALSE;
gboolean stereo_supported = FALSE;
fd = open ("/dev/dsp", O_RDWR);
if (fd < 0) {
perror ("/dev/dsp");
@ -88,29 +64,7 @@ main (int argc, char *argv[])
probe = g_new0 (Probe, 1);
probe->fd = fd;
probe->format = AFMT_S16_LE;
/* check if the device supports mono, stereo or both */
probe->n_channels = 1;
ret = ioctl (fd, SNDCTL_DSP_CHANNELS, &probe->n_channels);
if (probe->n_channels == 1)
mono_supported = TRUE;
probe->n_channels = 2;
ret = ioctl (fd, SNDCTL_DSP_CHANNELS, &probe->n_channels);
if (probe->n_channels == 2)
stereo_supported = TRUE;
if (mono_supported && stereo_supported) {
g_print ("The device supports mono and stereo.\n");
} else if (mono_supported) {
g_print ("The device supports only mono.\n");
} else if (stereo_supported) {
g_print ("The device supports only stereo.\n");
} else {
/* exit with error */
g_error
("The device doesn't support mono or stereo. This should not happen.\n");
}
probe_check (probe);
g_array_sort (probe->rates, int_compare);