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oss: port to 0.11

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This commit is contained in:
Mark Nauwelaerts 2012-04-20 18:12:54 +02:00
parent ad5c3cd3dd
commit c4c3736e1a
17 changed files with 205 additions and 1256 deletions
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2
configure.ac
View file

@ -310,7 +310,7 @@ dnl Non ported plugins (non-dependant, then dependant)
dnl Make sure you have a space before and after all plugins
GST_PLUGINS_NONPORTED="deinterlace \
cairo cairo_gobject gdk_pixbuf \
oss oss4 \
oss4 \
osx_video osx_audio "
AC_SUBST(GST_PLUGINS_NONPORTED)

1
docs/plugins/Makefile.am
View file

@ -194,7 +194,6 @@ EXTRA_HFILES = \
$(top_srcdir)/sys/oss4/oss4-mixer.h \
$(top_srcdir)/sys/oss4/oss4-sink.h \
$(top_srcdir)/sys/oss4/oss4-source.h \
$(top_srcdir)/sys/oss/gstossmixerelement.h \
$(top_srcdir)/sys/oss/gstosssink.h \
$(top_srcdir)/sys/oss/gstosssrc.h \
$(top_srcdir)/sys/osxaudio/gstosxaudiosrc.h \

1
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -114,7 +114,6 @@
<xi:include href="xml/element-oss4mixer.xml" />
<xi:include href="xml/element-oss4sink.xml" />
<xi:include href="xml/element-oss4src.xml" />
<xi:include href="xml/element-ossmixer.xml" />
<xi:include href="xml/element-osssink.xml" />
<xi:include href="xml/element-osssrc.xml" />
<xi:include href="xml/element-osxaudiosink.xml" />

14
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -1457,20 +1457,6 @@ GstOss4SourceInput
GstOss4SourceInputClass
</SECTION>
<SECTION>
<FILE>element-ossmixer</FILE>
<TITLE>ossmixer</TITLE>
GstOssMixerElement
<SUBSECTION Standard>
GstOssMixerElementClass
GST_OSS_MIXER_ELEMENT
GST_OSS_MIXER_ELEMENT_CLASS
GST_IS_OSS_MIXER_ELEMENT
GST_IS_OSS_MIXER_ELEMENT_CLASS
GST_TYPE_OSS_MIXER_ELEMENT
gst_oss_mixer_element_get_type
</SECTION>
<SECTION>
<FILE>element-osssink</FILE>
<TITLE>osssink</TITLE>

9
sys/oss/Makefile.am
View file

@ -2,16 +2,12 @@ plugin_LTLIBRARIES = libgstossaudio.la
libgstossaudio_la_SOURCES = gstossaudio.c \
gstosshelper.c \
gstossmixer.c \
gstossmixerelement.c \
gstossmixertrack.c \
gstosssink.c \
gstosssrc.c
libgstossaudio_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstossaudio_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) \
-lgstinterfaces-$(GST_API_VERSION) \
-lgstaudio-$(GST_API_VERSION) \
$(GST_BASE_LIBS) \
$(GST_LIBS)
@ -22,7 +18,4 @@ noinst_HEADERS = common.h \
gstosssink.h \
gstosssrc.h \
gstosshelper.h \
gstossdmabuffer.h \
gstossmixer.h \
gstossmixerelement.h \
gstossmixertrack.h
gstossdmabuffer.h

5
sys/oss/gstossaudio.c
View file

@ -23,7 +23,6 @@
#include "gst/gst-i18n-plugin.h"
#include "gstossmixerelement.h"
#include "gstosssink.h"
#include "gstosssrc.h"
@ -33,9 +32,7 @@ GST_DEBUG_CATEGORY (oss_debug);
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "ossmixer", GST_RANK_NONE,
GST_TYPE_OSS_MIXER_ELEMENT) ||
!gst_element_register (plugin, "osssrc", GST_RANK_SECONDARY,
if (!gst_element_register (plugin, "osssrc", GST_RANK_SECONDARY,
GST_TYPE_OSS_SRC) ||
!gst_element_register (plugin, "osssink", GST_RANK_SECONDARY,
GST_TYPE_OSSSINK)) {

52
sys/oss/gstosshelper.c
View file

@ -47,10 +47,7 @@
# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include <gst/interfaces/propertyprobe.h>
#include "gstosshelper.h"
#include "gstossmixer.h"
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
@ -176,19 +173,19 @@ gst_oss_helper_get_format_structure (unsigned int format_bit)
format = "U8";
break;
case AFMT_S16_LE:
format = "S16_LE";
format = "S16LE";
break;
case AFMT_S16_BE:
format = "S16_BE";
format = "S16BE";
break;
case AFMT_S8:
format = "S8";
break;
case AFMT_U16_LE:
format = "U16_LE";
format = "U16LE";
break;
case AFMT_U16_BE:
format = "U16_BE";
format = "U16BE";
break;
default:
g_assert_not_reached ();
@ -196,7 +193,8 @@ gst_oss_helper_get_format_structure (unsigned int format_bit)
}
structure = gst_structure_new ("audio/x-raw",
"format", G_TYPE_STRING, format, NULL);
"format", G_TYPE_STRING, format,
"layout", G_TYPE_STRING, "interleaved", NULL);
return structure;
}
@ -386,3 +384,41 @@ gst_oss_helper_rate_int_compare (gconstpointer a, gconstpointer b)
return 1;
return 0;
}
gchar *
gst_oss_helper_get_card_name (const gchar * mixer_name)
{
#ifdef SOUND_MIXER_INFO
struct mixer_info minfo;
#endif
gint fd;
gchar *name = NULL;
GST_INFO ("Opening mixer for device %s", mixer_name);
fd = open (mixer_name, O_RDWR);
if (fd == -1)
goto open_failed;
/* get name, not fatal */
#ifdef SOUND_MIXER_INFO
if (ioctl (fd, SOUND_MIXER_INFO, &minfo) == 0) {
name = g_strdup (minfo.name);
GST_INFO ("Card name = %s", GST_STR_NULL (name));
} else
#endif
{
name = g_strdup ("Unknown");
GST_INFO ("Unknown card name");
}
return name;
/* ERRORS */
open_failed:
{
/* this is valid. OSS devices don't need to expose a mixer */
GST_DEBUG ("Failed to open mixer device %s, mixing disabled: %s",
mixer_name, strerror (errno));
return NULL;
}
}

3
sys/oss/gstosshelper.h
View file

@ -36,6 +36,9 @@ G_BEGIN_DECLS
GstCaps* gst_oss_helper_probe_caps (gint fd);
gchar *gst_oss_helper_get_card_name (const gchar * mixer_name);
G_END_DECLS

378
sys/oss/gstossmixer.c
View file

@ -1,378 +0,0 @@
/* GStreamer OSS Mixer implementation
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* gstossmixer.c: mixer interface implementation for OSS
*
* 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 <stdio.h>
#include <stdlib.h>
#include <fcntl.h>
#include <unistd.h>
#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
# ifdef HAVE_OSS_INCLUDE_IN_MACHINE
# include <machine/soundcard.h>
# else
# error "What to include?"
# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include <gst/gst-i18n-plugin.h>
#include "gstossmixer.h"
#include "gstossmixertrack.h"
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
#define MASK_BIT_IS_SET(mask, bit) \
(mask & (1 << bit))
static gboolean
gst_ossmixer_open (GstOssMixer * mixer)
{
#ifdef SOUND_MIXER_INFO
struct mixer_info minfo;
#endif
g_return_val_if_fail (mixer->mixer_fd == -1, FALSE);
mixer->mixer_fd = open (mixer->device, O_RDWR);
if (mixer->mixer_fd == -1)
goto open_failed;
/* get masks */
if (ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECMASK, &mixer->recmask) < 0
|| ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECSRC, &mixer->recdevs) < 0
|| ioctl (mixer->mixer_fd, SOUND_MIXER_READ_STEREODEVS,
&mixer->stereomask) < 0
|| ioctl (mixer->mixer_fd, SOUND_MIXER_READ_DEVMASK, &mixer->devmask) < 0
|| ioctl (mixer->mixer_fd, SOUND_MIXER_READ_CAPS, &mixer->mixcaps) < 0)
goto masks_failed;
/* get name, not fatal */
g_free (mixer->cardname);
#ifdef SOUND_MIXER_INFO
if (ioctl (mixer->mixer_fd, SOUND_MIXER_INFO, &minfo) == 0) {
mixer->cardname = g_strdup (minfo.name);
GST_INFO ("Card name = %s", GST_STR_NULL (mixer->cardname));
} else
#endif
{
mixer->cardname = g_strdup ("Unknown");
GST_INFO ("Unknown card name");
}
GST_INFO ("Opened mixer for device %s", mixer->device);
return TRUE;
/* ERRORS */
open_failed:
{
/* this is valid. OSS devices don't need to expose a mixer */
GST_DEBUG ("Failed to open mixer device %s, mixing disabled: %s",
mixer->device, strerror (errno));
return FALSE;
}
masks_failed:
{
GST_DEBUG ("Failed to get device masks");
close (mixer->mixer_fd);
mixer->mixer_fd = -1;
return FALSE;
}
}
static void
gst_ossmixer_ensure_track_list (GstOssMixer * mixer)
{
gint i, master = -1;
g_return_if_fail (mixer->mixer_fd != -1);
if (mixer->tracklist)
return;
/* find master volume */
if (mixer->devmask & SOUND_MASK_VOLUME)
master = SOUND_MIXER_VOLUME;
else if (mixer->devmask & SOUND_MASK_PCM)
master = SOUND_MIXER_PCM;
else if (mixer->devmask & SOUND_MASK_SPEAKER)
master = SOUND_MIXER_SPEAKER; /* doubtful... */
/* else: no master, so we won't set any */
/* build track list */
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (mixer->devmask & (1 << i)) {
GstMixerTrack *track;
gboolean input = FALSE, stereo = FALSE, record = FALSE;
/* track exists, make up capabilities */
if (MASK_BIT_IS_SET (mixer->stereomask, i))
stereo = TRUE;
if (MASK_BIT_IS_SET (mixer->recmask, i))
input = TRUE;
if (MASK_BIT_IS_SET (mixer->recdevs, i))
record = TRUE;
/* do we want this in our list? */
if (!((mixer->dir & GST_OSS_MIXER_CAPTURE && input == TRUE) ||
(mixer->dir & GST_OSS_MIXER_PLAYBACK && i != SOUND_MIXER_PCM)))
/* the PLAYBACK case seems hacky, but that's how 0.8 had it */
continue;
/* add track to list */
track = gst_ossmixer_track_new (mixer->mixer_fd, i, stereo ? 2 : 1,
(record ? GST_MIXER_TRACK_RECORD : 0) |
(input ? GST_MIXER_TRACK_INPUT :
GST_MIXER_TRACK_OUTPUT) |
((master != i) ? 0 : GST_MIXER_TRACK_MASTER));
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
}
}
GstOssMixer *
gst_ossmixer_new (const char *device, GstOssMixerDirection dir)
{
GstOssMixer *ret = NULL;
g_return_val_if_fail (device != NULL, NULL);
ret = g_new0 (GstOssMixer, 1);
ret->device = g_strdup (device);
ret->dir = dir;
ret->mixer_fd = -1;
if (!gst_ossmixer_open (ret))
goto error;
return ret;
/* ERRORS */
error:
{
gst_ossmixer_free (ret);
return NULL;
}
}
void
gst_ossmixer_free (GstOssMixer * mixer)
{
g_return_if_fail (mixer != NULL);
if (mixer->device) {
g_free (mixer->device);
mixer->device = NULL;
}
if (mixer->cardname) {
g_free (mixer->cardname);
mixer->cardname = NULL;
}
if (mixer->tracklist) {
g_list_foreach (mixer->tracklist, (GFunc) g_object_unref, NULL);
g_list_free (mixer->tracklist);
mixer->tracklist = NULL;
}
if (mixer->mixer_fd != -1) {
close (mixer->mixer_fd);
mixer->mixer_fd = -1;
}
g_free (mixer);
}
/* unused with G_DISABLE_* */
static G_GNUC_UNUSED gboolean
gst_ossmixer_contains_track (GstOssMixer * mixer, GstOssMixerTrack * osstrack)
{
const GList *item;
for (item = mixer->tracklist; item != NULL; item = item->next)
if (item->data == osstrack)
return TRUE;
return FALSE;
}
const GList *
gst_ossmixer_list_tracks (GstOssMixer * mixer)
{
gst_ossmixer_ensure_track_list (mixer);
return (const GList *) mixer->tracklist;
}
void
gst_ossmixer_get_volume (GstOssMixer * mixer,
GstMixerTrack * track, gint * volumes)
{
gint volume;
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
g_return_if_fail (mixer->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
if (track->flags & GST_MIXER_TRACK_MUTE) {
volumes[0] = osstrack->lvol;
if (track->num_channels == 2) {
volumes[1] = osstrack->rvol;
}
} else {
/* get */
if (ioctl (mixer->mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
g_warning ("Error getting recording device (%d) volume: %s",
osstrack->track_num, strerror (errno));
volume = 0;
}
osstrack->lvol = volumes[0] = (volume & 0xff);
if (track->num_channels == 2) {
osstrack->rvol = volumes[1] = ((volume >> 8) & 0xff);
}
}
}
void
gst_ossmixer_set_volume (GstOssMixer * mixer,
GstMixerTrack * track, gint * volumes)
{
gint volume;
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
g_return_if_fail (mixer->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
/* prepare the value for ioctl() */
if (!(track->flags & GST_MIXER_TRACK_MUTE)) {
volume = (volumes[0] & 0xff);
if (track->num_channels == 2) {
volume |= ((volumes[1] & 0xff) << 8);
}
/* set */
if (ioctl (mixer->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
g_warning ("Error setting recording device (%d) volume (0x%x): %s",
osstrack->track_num, volume, strerror (errno));
return;
}
}
osstrack->lvol = volumes[0];
if (track->num_channels == 2) {
osstrack->rvol = volumes[1];
}
}
void
gst_ossmixer_set_mute (GstOssMixer * mixer, GstMixerTrack * track,
gboolean mute)
{
int volume;
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
g_return_if_fail (mixer->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
if (mute) {
volume = 0;
} else {
volume = (osstrack->lvol & 0xff);
if (MASK_BIT_IS_SET (mixer->stereomask, osstrack->track_num)) {
volume |= ((osstrack->rvol & 0xff) << 8);
}
}
if (ioctl (mixer->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
g_warning ("Error setting mixer recording device volume (0x%x): %s",
volume, strerror (errno));
return;
}
if (mute) {
track->flags |= GST_MIXER_TRACK_MUTE;
} else {
track->flags &= ~GST_MIXER_TRACK_MUTE;
}
}
void
gst_ossmixer_set_record (GstOssMixer * mixer,
GstMixerTrack * track, gboolean record)
{
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
g_return_if_fail (mixer->mixer_fd != -1);
g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
/* if there's nothing to do... */
if ((record && GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) ||
(!record && !GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)))
return;
/* if we're exclusive, then we need to unset the current one(s) */
if (mixer->mixcaps & SOUND_CAP_EXCL_INPUT) {
GList *track;
for (track = mixer->tracklist; track != NULL; track = track->next) {
GstMixerTrack *turn = (GstMixerTrack *) track->data;
turn->flags &= ~GST_MIXER_TRACK_RECORD;
}
mixer->recdevs = 0;
}
/* set new record bit, if needed */
if (record) {
mixer->recdevs |= (1 << osstrack->track_num);
} else {
mixer->recdevs &= ~(1 << osstrack->track_num);
}
/* set it to the device */
if (ioctl (mixer->mixer_fd, SOUND_MIXER_WRITE_RECSRC, &mixer->recdevs) < 0) {
g_warning ("Error setting mixer recording devices (0x%x): %s",
mixer->recdevs, strerror (errno));
return;
}
if (record) {
track->flags |= GST_MIXER_TRACK_RECORD;
} else {
track->flags &= ~GST_MIXER_TRACK_RECORD;
}
}

171
sys/oss/gstossmixer.h
View file

@ -1,171 +0,0 @@
/* GStreamer OSS Mixer implementation
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* gstossmixer.h: mixer interface implementation for OSS
*
* 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_OSS_MIXER_H__
#define __GST_OSS_MIXER_H__
#include <gst/gst.h>
#include <gst/interfaces/mixer.h>
#include "gstosshelper.h"
G_BEGIN_DECLS
#define GST_OSS_MIXER(obj) ((GstOssMixer*)(obj))
typedef enum {
GST_OSS_MIXER_CAPTURE = 1<<0,
GST_OSS_MIXER_PLAYBACK = 1<<1,
GST_OSS_MIXER_ALL = GST_OSS_MIXER_CAPTURE | GST_OSS_MIXER_PLAYBACK
} GstOssMixerDirection;
typedef struct _GstOssMixer GstOssMixer;
struct _GstOssMixer {
GList * tracklist; /* list of available tracks */
gint mixer_fd;
gchar * device;
gchar * cardname;
gint recmask;
gint recdevs;
gint stereomask;
gint devmask;
gint mixcaps;
GstOssMixerDirection dir;
};
GstOssMixer* gst_ossmixer_new (const gchar *device,
GstOssMixerDirection dir);
void gst_ossmixer_free (GstOssMixer *mixer);
const GList* gst_ossmixer_list_tracks (GstOssMixer * mixer);
void gst_ossmixer_set_volume (GstOssMixer * mixer,
GstMixerTrack * track,
gint * volumes);
void gst_ossmixer_get_volume (GstOssMixer * mixer,
GstMixerTrack * track,
gint * volumes);
void gst_ossmixer_set_record (GstOssMixer * mixer,
GstMixerTrack * track,
gboolean record);
void gst_ossmixer_set_mute (GstOssMixer * mixer,
GstMixerTrack * track,
gboolean mute);
#define GST_IMPLEMENT_OSS_MIXER_METHODS(Type, interface_as_function) \
static gboolean \
interface_as_function ## _supported (Type *this, GType iface_type) \
{ \
g_assert (iface_type == GST_TYPE_MIXER); \
\
return (this->mixer != NULL); \
} \
\
static const GList* \
interface_as_function ## _list_tracks (GstMixer * mixer) \
{ \
Type *this = (Type*) mixer; \
\
g_return_val_if_fail (this != NULL, NULL); \
g_return_val_if_fail (this->mixer != NULL, NULL); \
\
return gst_ossmixer_list_tracks (this->mixer); \
} \
\
static void \
interface_as_function ## _set_volume (GstMixer * mixer, GstMixerTrack * track, \
gint * volumes) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_ossmixer_set_volume (this->mixer, track, volumes); \
} \
\
static void \
interface_as_function ## _get_volume (GstMixer * mixer, GstMixerTrack * track, \
gint * volumes) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_ossmixer_get_volume (this->mixer, track, volumes); \
} \
\
static void \
interface_as_function ## _set_record (GstMixer * mixer, GstMixerTrack * track, \
gboolean record) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_ossmixer_set_record (this->mixer, track, record); \
} \
\
static void \
interface_as_function ## _set_mute (GstMixer * mixer, GstMixerTrack * track, \
gboolean mute) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_ossmixer_set_mute (this->mixer, track, mute); \
} \
\
static void \
interface_as_function ## _interface_init (GstMixerInterface * iface) \
{ \
GST_MIXER_TYPE (iface) = GST_MIXER_HARDWARE; \
\
/* set up the interface hooks */ \
iface->list_tracks = interface_as_function ## _list_tracks; \
iface->set_volume = interface_as_function ## _set_volume; \
iface->get_volume = interface_as_function ## _get_volume; \
iface->set_mute = interface_as_function ## _set_mute; \
iface->set_record = interface_as_function ## _set_record; \
}
G_END_DECLS
#endif /* __GST_OSS_MIXER_H__ */

218
sys/oss/gstossmixerelement.c
View file

@ -1,218 +0,0 @@
/* OSS mixer interface element.
* Copyright (C) 2005 Andrew Vander Wingo <wingo@pobox.com>
*
* 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.
*/
/**
* SECTION:element-ossmixer
*
* This element lets you adjust sound input and output levels with the
* Open Sound System (OSS). It supports the #GstMixer interface, which can be
* used to obtain a list of available mixer tracks. Set the mixer element to
* READY state before using the #GstMixer interface on it.
*
* <refsect2>
* <title>Example pipelines</title>
* <para>
* ossmixer can't be used in a sensible way in gst-launch.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstossmixerelement.h"
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
#define DEFAULT_DEVICE "/dev/mixer"
#define DEFAULT_DEVICE_NAME NULL
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME
};
GST_BOILERPLATE_WITH_INTERFACE (GstOssMixerElement, gst_oss_mixer_element,
GstElement, GST_TYPE_ELEMENT, GstMixer, GST_TYPE_MIXER,
gst_oss_mixer_element);
GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssMixerElement, gst_oss_mixer_element);
static GstStateChangeReturn gst_oss_mixer_element_change_state (GstElement *
element, GstStateChange transition);
static void gst_oss_mixer_element_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_oss_mixer_element_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static void gst_oss_mixer_element_finalize (GObject * object);
static void
gst_oss_mixer_element_base_init (gpointer klass)
{
gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass), "OSS Mixer",
"Generic/Audio",
"Control sound input and output levels with OSS",
"Andrew Vander Wingo <wingo@pobox.com>");
}
static void
gst_oss_mixer_element_class_init (GstOssMixerElementClass * klass)
{
GstElementClass *element_class;
GObjectClass *gobject_class;
element_class = (GstElementClass *) klass;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = gst_oss_mixer_element_finalize;
gobject_class->set_property = gst_oss_mixer_element_set_property;
gobject_class->get_property = gst_oss_mixer_element_get_property;
/**
* GstOssMixerElement:device
*
* OSS mixer device (usually /dev/mixer)
*
* Since: 0.10.5
**/
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"OSS mixer device (usually /dev/mixer)", DEFAULT_DEVICE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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 | G_PARAM_STATIC_STRINGS));
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_oss_mixer_element_change_state);
}
static void
gst_oss_mixer_element_finalize (GObject * obj)
{
GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (obj);
g_free (this->device);
G_OBJECT_CLASS (parent_class)->finalize (obj);
}
static void
gst_oss_mixer_element_init (GstOssMixerElement * this,
GstOssMixerElementClass * g_class)
{
this->mixer = NULL;
this->device = g_strdup (DEFAULT_DEVICE);
}
static void
gst_oss_mixer_element_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (this->device);
this->device = g_value_dup_string (value);
/* make sure we never set NULL */
if (this->device == NULL) {
this->device = g_strdup (DEFAULT_DEVICE);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_oss_mixer_element_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, this->device);
break;
case PROP_DEVICE_NAME:
if (this->mixer) {
g_value_set_string (value, this->mixer->cardname);
} else {
g_value_set_string (value, NULL);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_oss_mixer_element_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!this->mixer) {
this->mixer = gst_ossmixer_new (this->device, GST_OSS_MIXER_ALL);
if (!this->mixer)
goto open_failed;
}
break;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mixer) {
gst_ossmixer_free (this->mixer);
this->mixer = NULL;
}
break;
default:
break;
}
return ret;
/* ERRORS */
open_failed:
{
GST_ELEMENT_ERROR (element, RESOURCE, OPEN_READ_WRITE, (NULL),
("Failed to open oss mixer device '%s'", this->device));
return GST_STATE_CHANGE_FAILURE;
}
}

59
sys/oss/gstossmixerelement.h
View file

@ -1,59 +0,0 @@
/* OSS mixer interface element.
* Copyright (C) 2005 Andrew Vander Wingo <wingo@pobox.com>
*
* 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., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __GST_OSS_MIXER_ELEMENT_H__
#define __GST_OSS_MIXER_ELEMENT_H__
#include "gstossmixer.h"
G_BEGIN_DECLS
#define GST_OSS_MIXER_ELEMENT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElement))
#define GST_OSS_MIXER_ELEMENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElementClass))
#define GST_IS_OSS_MIXER_ELEMENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS_MIXER_ELEMENT))
#define GST_IS_OSS_MIXER_ELEMENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS_MIXER_ELEMENT))
#define GST_TYPE_OSS_MIXER_ELEMENT (gst_oss_mixer_element_get_type())
typedef struct _GstOssMixerElement GstOssMixerElement;
typedef struct _GstOssMixerElementClass GstOssMixerElementClass;
struct _GstOssMixerElement {
GstElement parent;
gchar *device;
GstOssMixer *mixer;
};
struct _GstOssMixerElementClass {
GstElementClass parent;
};
GType gst_oss_mixer_element_get_type (void);
G_END_DECLS
#endif /* __GST_OSS_MIXER_ELEMENT_H__ */

172
sys/oss/gstossmixertrack.c
View file

@ -1,172 +0,0 @@
/* GStreamer OSS Mixer implementation
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* gstossmixer.c: mixer interface implementation for OSS
*
* 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 <stdio.h>
#include <stdlib.h>
#include <fcntl.h>
#include <unistd.h>
#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
# ifdef HAVE_OSS_INCLUDE_IN_MACHINE
# include <machine/soundcard.h>
# else
# error "What to include?"
# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include <gst/gst-i18n-plugin.h>
#include "gstossmixertrack.h"
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
#define MASK_BIT_IS_SET(mask, bit) \
(mask & (1 << bit))
G_DEFINE_TYPE (GstOssMixerTrack, gst_ossmixer_track, GST_TYPE_MIXER_TRACK);
static void
gst_ossmixer_track_class_init (GstOssMixerTrackClass * klass)
{
/* nop */
}
static void
gst_ossmixer_track_init (GstOssMixerTrack * track)
{
track->lvol = track->rvol = 0;
track->track_num = 0;
}
static const gchar **labels = NULL;
/* three functions: firstly, OSS has the nasty habit of inserting
* spaces in the labels, we want to get rid of them. Secondly,
* i18n is impossible with OSS' way of providing us with mixer
* labels, so we make a 'given' list of i18n'ed labels. Thirdly, I
* personally don't like the "1337" names that OSS gives to their
* labels ("Vol", "Mic", "Rec"), I'd rather see full names. */
static void
fill_labels (void)
{
gint i, pos;
const gchar *origs[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_LABELS;
const struct
{
const gchar *given;
const gchar *wanted;
}
cases[] = {
/* Note: this list is simply ripped from soundcard.h. For
* some people, some values might be missing (3D surround,
* etc.) - feel free to add them. That's the reason why
* I'm doing this in such a horribly complicated way. */
{
"Vol ", _("Volume")}, {
"Bass ", _("Bass")}, {
"Trebl", _("Treble")}, {
"Synth", _("Synth")}, {
"Pcm ", _("PCM")}, {
"Spkr ", _("Speaker")}, {
"Line ", _("Line-in")}, {
"Mic ", _("Microphone")}, {
"CD ", _("CD")}, {
"Mix ", _("Mixer")}, {
"Pcm2 ", _("PCM-2")}, {
"Rec ", _("Record")}, {
"IGain", _("In-gain")}, {
"OGain", _("Out-gain")}, {
"Line1", _("Line-1")}, {
"Line2", _("Line-2")}, {
"Line3", _("Line-3")}, {
"Digital1", _("Digital-1")}, {
"Digital2", _("Digital-2")}, {
"Digital3", _("Digital-3")}, {
"PhoneIn", _("Phone-in")}, {
"PhoneOut", _("Phone-out")}, {
"Video", _("Video")}, {
"Radio", _("Radio")}, {
"Monitor", _("Monitor")}, {
NULL, NULL}
};
labels = g_malloc (sizeof (gchar *) * SOUND_MIXER_NRDEVICES);
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
for (pos = 0; cases[pos].given != NULL; pos++) {
if (!strcmp (cases[pos].given, origs[i])) {
labels[i] = g_strdup (cases[pos].wanted);
break;
}
}
if (cases[pos].given == NULL)
labels[i] = g_strdup (origs[i]);
}
}
GstMixerTrack *
gst_ossmixer_track_new (gint mixer_fd,
gint track_num, gint max_chans, gint flags)
{
GstOssMixerTrack *osstrack;
GstMixerTrack *track;
gint volume;
if (!labels)
fill_labels ();
osstrack = g_object_new (GST_TYPE_OSSMIXER_TRACK, NULL);
track = GST_MIXER_TRACK (osstrack);
track->label = g_strdup (labels[track_num]);
track->num_channels = max_chans;
track->flags = flags;
track->min_volume = 0;
track->max_volume = 100;
osstrack->track_num = track_num;
/* volume */
if (ioctl (mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
g_warning ("Error getting device (%d) volume: %s",
osstrack->track_num, strerror (errno));
volume = 0;
}
osstrack->lvol = (volume & 0xff);
if (track->num_channels == 2) {
osstrack->rvol = ((volume >> 8) & 0xff);
}
return track;
}

62
sys/oss/gstossmixertrack.h
View file

@ -1,62 +0,0 @@
/* GStreamer OSS Mixer implementation
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* gstossmixertrack.h: OSS mixer tracks
*
* 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_OSS_MIXER_TRACK_H__
#define __GST_OSS_MIXER_TRACK_H__
#include <gst/gst.h>
#include <gst/interfaces/mixer.h>
#include "gstosshelper.h"
G_BEGIN_DECLS
#define GST_TYPE_OSSMIXER_TRACK \
(gst_ossmixer_track_get_type ())
#define GST_OSSMIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_OSSMIXER_TRACK, \
GstOssMixerTrack))
#define GST_OSSMIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_OSSMIXER_TRACK, \
GstOssMixerTrackClass))
#define GST_IS_OSSMIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_OSSMIXER_TRACK))
#define GST_IS_OSSMIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_OSSMIXER_TRACK))
typedef struct _GstOssMixerTrack {
GstMixerTrack parent;
gint lvol, rvol;
gint track_num;
} GstOssMixerTrack;
typedef struct _GstOssMixerTrackClass {
GstMixerTrackClass parent;
} GstOssMixerTrackClass;
GType gst_ossmixer_track_get_type (void);
GstMixerTrack* gst_ossmixer_track_new (gint mixer_fd,
gint track_num, gint max_chans, gint flags);
G_END_DECLS
#endif /* __GST_OSS_MIXER_TRACK_H__ */

161
sys/oss/gstosssink.c
View file

@ -73,10 +73,6 @@
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
static void gst_oss_sink_base_init (gpointer g_class);
static void gst_oss_sink_class_init (GstOssSinkClass * klass);
static void gst_oss_sink_init (GstOssSink * osssink);
static void gst_oss_sink_dispose (GObject * object);
static void gst_oss_sink_finalise (GObject * object);
@ -85,14 +81,14 @@ static void gst_oss_sink_get_property (GObject * object, guint prop_id,
static void gst_oss_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink);
static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter);
static gboolean gst_oss_sink_open (GstAudioSink * asink);
static gboolean gst_oss_sink_close (GstAudioSink * asink);
static gboolean gst_oss_sink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
GstAudioRingBufferSpec * spec);
static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);
static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
static gint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_oss_sink_delay (GstAudioSink * asink);
static void gst_oss_sink_reset (GstAudioSink * asink);
@ -128,35 +124,10 @@ static GstStaticPadTemplate osssink_sink_factory =
"channels = (int) 2, " "channel-mask = (bitmask) 0x3")
);
static GstElementClass *parent_class = NULL;
/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_oss_sink_get_type (void)
{
static GType osssink_type = 0;
if (!osssink_type) {
static const GTypeInfo osssink_info = {
sizeof (GstOssSinkClass),
gst_oss_sink_base_init,
NULL,
(GClassInitFunc) gst_oss_sink_class_init,
NULL,
NULL,
sizeof (GstOssSink),
0,
(GInstanceInitFunc) gst_oss_sink_init,
};
osssink_type =
g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink",
&osssink_info, 0);
}
return osssink_type;
}
#define gst_oss_sink_parent_class parent_class
G_DEFINE_TYPE (GstOssSink, gst_oss_sink, GST_TYPE_AUDIO_SINK);
static void
gst_oss_sink_dispose (GObject * object)
@ -171,29 +142,16 @@ gst_oss_sink_dispose (GObject * object)
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_oss_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_static_metadata (element_class, "Audio Sink (OSS)",
"Sink/Audio",
"Output to a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Wim Taymans <wim.taymans@chello.be>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&osssink_sink_factory));
}
static void
gst_oss_sink_class_init (GstOssSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
@ -218,6 +176,15 @@ gst_oss_sink_class_init (GstOssSinkClass * klass)
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);
gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (OSS)",
"Sink/Audio",
"Output to a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Wim Taymans <wim.taymans@chello.be>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&osssink_sink_factory));
}
static void
@ -286,7 +253,7 @@ gst_oss_sink_get_property (GObject * object, guint prop_id,
}
static GstCaps *
gst_oss_sink_getcaps (GstBaseSink * bsink)
gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
{
GstOssSink *osssink;
GstCaps *caps;
@ -304,7 +271,16 @@ gst_oss_sink_getcaps (GstBaseSink * bsink)
}
}
return caps;
if (filter && caps) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
return intersection;
} else {
return caps;
}
}
static gint
@ -325,41 +301,50 @@ ilog2 (gint x)
}
static gint
gst_oss_sink_get_format (GstBufferFormat fmt)
gst_oss_sink_get_format (GstAudioRingBufferFormatType fmt, GstAudioFormat rfmt)
{
gint result;
switch (fmt) {
case GST_MU_LAW:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
result = AFMT_MU_LAW;
break;
case GST_A_LAW:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
result = AFMT_A_LAW;
break;
case GST_IMA_ADPCM:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_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:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
result = AFMT_MPEG;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
{
switch (rfmt) {
case GST_AUDIO_FORMAT_U8:
result = AFMT_U8;
break;
case GST_AUDIO_FORMAT_S16LE:
result = AFMT_S16_LE;
break;
case GST_AUDIO_FORMAT_S16BE:
result = AFMT_S16_BE;
break;
case GST_AUDIO_FORMAT_S8:
result = AFMT_S8;
break;
case GST_AUDIO_FORMAT_U16LE:
result = AFMT_U16_LE;
break;
case GST_AUDIO_FORMAT_U16BE:
result = AFMT_U16_BE;
break;
default:
result = 0;
break;
}
break;
}
default:
result = 0;
break;
@ -425,12 +410,13 @@ gst_oss_sink_close (GstAudioSink * asink)
}
static gboolean
gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
gst_oss_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
{
GstOssSink *oss;
struct audio_buf_info info;
int mode;
int tmp;
guint width, rate, channels;
oss = GST_OSSSINK (asink);
@ -446,18 +432,23 @@ gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
goto non_block;
}
tmp = gst_oss_sink_get_format (spec->format);
tmp = gst_oss_sink_get_format (spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info));
if (tmp == 0)
goto wrong_format;
if (spec->width != 16 && spec->width != 8)
width = GST_AUDIO_INFO_WIDTH (&spec->info);
rate = GST_AUDIO_INFO_RATE (&spec->info);
channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (width != 16 && width != 8)
goto dodgy_width;
SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT");
if (spec->channels == 2)
if (channels == 2)
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, channels, "CHANNELS");
SET_PARAM (oss, SNDCTL_DSP_SPEED, rate, "SPEED");
tmp = ilog2 (spec->segsize);
tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
@ -470,8 +461,7 @@ gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
spec->segsize = info.fragsize;
spec->segtotal = info.fragstotal;
spec->bytes_per_sample = (spec->width / 8) * spec->channels;
oss->bytes_per_sample = (spec->width / 8) * spec->channels;
oss->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
spec->segsize, spec->segtotal, tmp);
@ -489,13 +479,14 @@ non_block:
wrong_format:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("Unable to get format %d", spec->format));
("Unable to get format (%d, %d)", spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info)));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("unexpected width %d", spec->width));
("unexpected width %d", width));
return FALSE;
}
}
@ -526,7 +517,7 @@ couldnt_reopen:
}
}
static guint
static gint
gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
return write (GST_OSSSINK (asink)->fd, data, length);

150
sys/oss/gstosssrc.c
View file

@ -77,10 +77,8 @@ enum
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);
#define gst_oss_src_parent_class parent_class
G_DEFINE_TYPE (GstOssSrc, gst_oss_src, GST_TYPE_AUDIO_SRC);
static void gst_oss_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
@ -90,12 +88,12 @@ static void gst_oss_src_set_property (GObject * object, guint prop_id,
static void gst_oss_src_dispose (GObject * object);
static void gst_oss_src_finalize (GstOssSrc * osssrc);
static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc);
static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
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);
GstAudioRingBufferSpec * 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);
@ -124,28 +122,16 @@ 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_static_metadata (element_class, "Audio Source (OSS)",
"Source/Audio",
"Capture from a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
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;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
@ -173,6 +159,15 @@ gst_oss_src_class_init (GstOssSrcClass * klass)
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (gstelement_class, "Audio Source (OSS)",
"Source/Audio",
"Capture from a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&osssrc_src_factory));
}
static void
@ -217,7 +212,7 @@ gst_oss_src_get_property (GObject * object, guint prop_id,
}
static void
gst_oss_src_init (GstOssSrc * osssrc, GstOssSrcClass * g_class)
gst_oss_src_init (GstOssSrc * osssrc)
{
const gchar *device;
@ -243,7 +238,7 @@ gst_oss_src_finalize (GstOssSrc * osssrc)
}
static GstCaps *
gst_oss_src_getcaps (GstBaseSrc * bsrc)
gst_oss_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstOssSrc *osssrc;
GstCaps *caps;
@ -268,7 +263,16 @@ gst_oss_src_getcaps (GstBaseSrc * bsrc)
GST_INFO_OBJECT (osssrc, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
if (filter && caps) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
return intersection;
} else {
return caps;
}
}
static gint
@ -289,41 +293,50 @@ ilog2 (gint x)
}
static gint
gst_oss_src_get_format (GstBufferFormat fmt)
gst_oss_src_get_format (GstAudioRingBufferFormatType fmt, GstAudioFormat rfmt)
{
gint result;
switch (fmt) {
case GST_MU_LAW:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
result = AFMT_MU_LAW;
break;
case GST_A_LAW:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
result = AFMT_A_LAW;
break;
case GST_IMA_ADPCM:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_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:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
result = AFMT_MPEG;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
{
switch (rfmt) {
case GST_AUDIO_FORMAT_U8:
result = AFMT_U8;
break;
case GST_AUDIO_FORMAT_S16LE:
result = AFMT_S16_LE;
break;
case GST_AUDIO_FORMAT_S16BE:
result = AFMT_S16_BE;
break;
case GST_AUDIO_FORMAT_S8:
result = AFMT_S8;
break;
case GST_AUDIO_FORMAT_U16LE:
result = AFMT_U16_LE;
break;
case GST_AUDIO_FORMAT_U16BE:
result = AFMT_U16_BE;
break;
default:
result = 0;
break;
}
break;
}
default:
result = 0;
break;
@ -352,14 +365,9 @@ gst_oss_src_open (GstAudioSrc * asrc)
}
}
if (!oss->mixer) {
oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
g_free (oss->device_name);
oss->device_name = gst_oss_helper_get_card_name ("/dev/mixer");
if (oss->mixer) {
g_free (oss->device_name);
oss->device_name = g_strdup (oss->mixer->cardname);
}
}
return TRUE;
no_permission:
@ -389,23 +397,19 @@ gst_oss_src_close (GstAudioSrc * asrc)
close (oss->fd);
if (oss->mixer) {
gst_ossmixer_free (oss->mixer);
oss->mixer = NULL;
}
gst_caps_replace (&oss->probed_caps, NULL);
return TRUE;
}
static gboolean
gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
gst_oss_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstOssSrc *oss;
struct audio_buf_info info;
int mode;
int fmt, tmp;
guint width, rate, channels;
oss = GST_OSS_SRC (asrc);
@ -414,10 +418,18 @@ gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
if (fcntl (oss->fd, F_SETFL, mode) == -1)
goto non_block;
fmt = gst_oss_src_get_format (spec->format);
fmt = gst_oss_src_get_format (spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info));
if (fmt == 0)
goto wrong_format;
width = GST_AUDIO_INFO_WIDTH (&spec->info);
rate = GST_AUDIO_INFO_RATE (&spec->info);
channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (width != 16 && width != 8)
goto dodgy_width;
tmp = ilog2 (spec->segsize);
tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
@ -428,22 +440,17 @@ gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
SET_PARAM (oss, SNDCTL_DSP_RESET, 0, "RESET");
SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt, "SETFMT");
if (spec->channels == 2)
if (channels == 2)
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, channels, "CHANNELS");
SET_PARAM (oss, SNDCTL_DSP_SPEED, rate, "SPEED");
GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE");
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);
oss->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
spec->segsize, spec->segtotal, tmp);
@ -460,13 +467,14 @@ non_block:
wrong_format:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unable to get format %d", spec->format), (NULL));
("Unable to get format (%d, %d)", spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info)), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unexpected width %d", spec->width), (NULL));
("Unexpected width %d", width), (NULL));
return FALSE;
}
}

3
sys/oss/gstosssrc.h
View file

@ -29,7 +29,6 @@
#include <gst/audio/gstaudiosrc.h>
#include "gstosshelper.h"
#include "gstossmixer.h"
G_BEGIN_DECLS
@ -52,8 +51,6 @@ struct _GstOssSrc {
gchar *device_name;
GstCaps *probed_caps;
GstOssMixer *mixer;
};
struct _GstOssSrcClass {