mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-19 06:46:38 +00:00
Code Issues 1 Projects Releases Wiki Activity

sys: remove sunaudio plugin

Browse source 
Even though hooked up to the build system, it's clear that no one
has ever built or used this with GStreamer 1.x. It wants to link
against libgstinterfaces, which no longer exists. And uses 0.10-style
raw audio caps. And the last meaningful change was done in 2009.
Let's just remove it.
This commit is contained in:
Tim-Philipp Müller 2017-06-23 19:52:04 +01:00
parent 2271fee77e
commit dd23afb6d4
24 changed files with 3 additions and 2858 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
README
View file

@ -138,7 +138,7 @@ PLATFORMS
- Linux is of course fully supported
- FreeBSD is reported to work; other BSDs should work too
- Solaris is reported to work; a specific sunaudiosink plugin has been written
- Solaris is reported to work
- MacOSX works, binary 1.x packages can be built using the cerbero build tool
- Windows works; binary 1.x packages can be built using the cerbero build tool
- MSys/MinGW builds

8
configure.ac
View file

@ -454,12 +454,6 @@ AG_GST_CHECK_FEATURE(OSS4, [Open Sound System 4], oss4, [
[OSS4_MISSING_HEADER="yes";HAVE_OSS4="no"])
])
dnl *** Sun Audio ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SUNAUDIO, true)
AG_GST_CHECK_FEATURE(SUNAUDIO, [Sun Audio], sunaudio, [
AC_CHECK_HEADER(sys/audioio.h, HAVE_SUNAUDIO="yes", HAVE_SUNAUDIO="no")
])
dnl *** OSX Audio ***
translit(dnm, m, l) AM_CONDITIONAL(USE_OSX_AUDIO, true)
HAVE_IOS="no"
@ -888,7 +882,6 @@ AM_CONDITIONAL(USE_PULSE, false)
AM_CONDITIONAL(USE_SHOUT2, false)
AM_CONDITIONAL(USE_SOUP, false)
AM_CONDITIONAL(USE_SPEEX, false)
AM_CONDITIONAL(USE_SUNAUDIO, false)
AM_CONDITIONAL(USE_TAGLIB, false)
AM_CONDITIONAL(USE_VPX, false)
AM_CONDITIONAL(USE_WAVEFORM, false)
@ -1035,7 +1028,6 @@ sys/oss/Makefile
sys/oss4/Makefile
sys/osxaudio/Makefile
sys/osxvideo/Makefile
sys/sunaudio/Makefile
sys/v4l2/Makefile
sys/waveform/Makefile
sys/ximage/Makefile

6
meson.build
View file

@ -100,16 +100,10 @@ check_functions = [
['HAVE_ISINF', 'isinf', '#include<math.h>'],
# check token HAVE_LIBV4L2
['HAVE_MMAP', 'mmap', '#include<sys/mman.h>'],
# check token HAVE_OSS
# check token HAVE_OSS4
# check token HAVE_OSS_INCLUDE_IN_MACHINE
# check token HAVE_OSS_INCLUDE_IN_ROOT
# check token HAVE_OSS_INCLUDE_IN_SYS
# check token HAVE_OSX_AUDIO
# check token HAVE_OSX_VIDEO
# check token HAVE_RDTSC
['HAVE_SINH', 'sinh', '#include<math.h>'],
# check token HAVE_SUNAUDIO
# check token HAVE_WAVEFORM
]

2
po/POTFILES.in
View file

@ -18,8 +18,6 @@ sys/oss4/oss4-source.c
sys/oss/gstosssink.c
sys/oss/gstosssrc.c
sys/osxaudio/gstosxaudioringbuffer.c
sys/sunaudio/gstsunaudiomixeroptions.c
sys/sunaudio/gstsunaudiomixertrack.c
sys/v4l2/gstv4l2bufferpool.c
sys/v4l2/gstv4l2object.c
sys/v4l2/gstv4l2radio.c

41
sys/Makefile.am
View file

@ -1,9 +1,3 @@
# if USE_DXR3
# DXR3_DIR=dxr3
# else
# DXR3_DIR=
# endif
if USE_OSS
OSS_DIR=oss
else
@ -16,7 +10,6 @@ else
OSS4_DIR=
endif
if USE_DIRECTSOUND
DIRECTSOUND_DIR=directsound
else
@ -29,12 +22,6 @@ else
WAVEFORM_DIR=
endif
if USE_SUNAUDIO
SUNAUDIO_DIR=sunaudio
else
SUNAUDIO_DIR=
endif
if USE_OSX_AUDIO
OSX_AUDIO_DIR=osxaudio
else
@ -47,44 +34,20 @@ else
OSX_VIDEO_DIR=
endif
# if USE_QCAM
# QCAM_DIR=qcam
# else
# QCAM_DIR=
# endif
if USE_GST_V4L2
V4L2_DIR=v4l2
else
V4L2_DIR=
endif
# if USE_VCD
# VCD_DIR=vcd
# else
# VCD_DIR=
# endif
# if USE_CDROM
# CDROM_DIR=cdrom
# else
# CDROM_DIR=
# endif
# if USE_OPENGL
# GL_DIR=glsink
# else
# GL_DIR=
# endif
if USE_X
XIMAGE_DIR=ximage
else
XIMAGE_DIR=
endif
SUBDIRS=$(DIRECTSOUND_DIR) $(WAVEFORM_DIR) $(OSS_DIR) $(OSS4_DIR) $(OSX_AUDIO_DIR) $(OSX_VIDEO_DIR) $(SUNAUDIO_DIR) $(V4L2_DIR) $(XIMAGE_DIR)
SUBDIRS=$(DIRECTSOUND_DIR) $(WAVEFORM_DIR) $(OSS_DIR) $(OSS4_DIR) $(OSX_AUDIO_DIR) $(OSX_VIDEO_DIR) $(V4L2_DIR) $(XIMAGE_DIR)
DIST_SUBDIRS=directsound oss oss4 osxaudio osxvideo sunaudio v4l2 waveform ximage
DIST_SUBDIRS=directsound oss oss4 osxaudio osxvideo v4l2 waveform ximage
include $(top_srcdir)/common/parallel-subdirs.mak

1
sys/meson.build
View file

@ -7,5 +7,4 @@ subdir('ximage')
# FIXME: Implement these
#subdir('osxaudio')
#subdir('osxvideo')
#subdir('sunaudio')
#subdir('waveform')

25
sys/sunaudio/Makefile.am
View file

@ -1,25 +0,0 @@
plugin_LTLIBRARIES = libgstsunaudio.la
libgstsunaudio_la_SOURCES = gstsunaudio.c \
gstsunaudiosink.c \
gstsunaudiomixerctrl.c \
gstsunaudiomixer.c \
gstsunaudiomixertrack.c \
gstsunaudiomixeroptions.c \
gstsunaudiosrc.c
libgstsunaudio_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
libgstsunaudio_la_LIBADD = \
-lgstinterfaces-@GST_API_VERSION@ \
-lgstaudio-@GST_API_VERSION@ \
$(GST_PLUGINS_BASE_LIBS) \
$(GST_LIBS)
libgstsunaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstsunaudiosink.h \
gstsunaudiomixer.h \
gstsunaudiomixerctrl.h \
gstsunaudiomixertrack.h \
gstsunaudiomixeroptions.h \
gstsunaudiosrc.h

62
sys/sunaudio/gstsunaudio.c
View file

@ -1,62 +0,0 @@
/*
* GStreamer - SunAudio plugin
* Copyright (C) 2005,2006 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gst/gst-i18n-plugin.h"
#include "gstsunaudiomixer.h"
#include "gstsunaudiosink.h"
#include "gstsunaudiosrc.h"
extern gchar *__gst_oss_plugin_dir;
GST_DEBUG_CATEGORY (sunaudio_debug);
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "sunaudiomixer", GST_RANK_NONE,
GST_TYPE_SUNAUDIO_MIXER) ||
!gst_element_register (plugin, "sunaudiosink", GST_RANK_SECONDARY,
GST_TYPE_SUNAUDIO_SINK) ||
!gst_element_register (plugin, "sunaudiosrc", GST_RANK_SECONDARY,
GST_TYPE_SUNAUDIO_SRC)) {
return FALSE;
}
GST_DEBUG_CATEGORY_INIT (sunaudio_debug, "sunaudio", 0, "sunaudio elements");
#ifdef ENABLE_NLS
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#endif /* ENABLE_NLS */
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
sunaudio,
"Sun Audio support for GStreamer",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

104
sys/sunaudio/gstsunaudiomixer.c
View file

@ -1,104 +0,0 @@
/*
* GStreamer - SunAudio mixer
* Copyright (C) 2005,2006 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-sunaudiomixer
*
* sunaudiomixer is an mixer that controls the sound input and output
* levels with the Sun Audio interface available in Solaris.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstsunaudiomixer.h"
GST_BOILERPLATE_WITH_INTERFACE (GstSunAudioMixer, gst_sunaudiomixer,
GstElement, GST_TYPE_ELEMENT, GstMixer, GST_TYPE_MIXER, gst_sunaudiomixer);
GST_IMPLEMENT_SUNAUDIO_MIXER_CTRL_METHODS (GstSunAudioMixer, gst_sunaudiomixer);
static GstStateChangeReturn gst_sunaudiomixer_change_state (GstElement *
element, GstStateChange transition);
static void
gst_sunaudiomixer_base_init (gpointer klass)
{
gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass),
"Sun Audio Mixer", "Generic/Audio",
"Control sound input and output levels with Sun Audio",
"Brian Cameron <brian.cameron@sun.com>");
}
static void
gst_sunaudiomixer_class_init (GstSunAudioMixerClass * klass)
{
GstElementClass *element_class;
element_class = (GstElementClass *) klass;
element_class->change_state = gst_sunaudiomixer_change_state;
}
static void
gst_sunaudiomixer_init (GstSunAudioMixer * this,
GstSunAudioMixerClass * g_class)
{
this->mixer = NULL;
}
static GstStateChangeReturn
gst_sunaudiomixer_change_state (GstElement * element, GstStateChange transition)
{
GstSunAudioMixer *this = GST_SUNAUDIO_MIXER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!this->mixer) {
const char *audiodev;
audiodev = g_getenv ("AUDIODEV");
if (audiodev == NULL) {
this->mixer = gst_sunaudiomixer_ctrl_new ("/dev/audioctl");
} else {
gchar *device = g_strdup_printf ("%sctl", audiodev);
this->mixer = gst_sunaudiomixer_ctrl_new (device);
g_free (device);
}
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mixer) {
gst_sunaudiomixer_ctrl_free (this->mixer);
this->mixer = NULL;
}
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}

51
sys/sunaudio/gstsunaudiomixer.h
View file

@ -1,51 +0,0 @@
/*
* GStreamer - SunAudio mixer
* Copyright (C) 2005,2006 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#ifndef __GST_SUNAUDIO_MIXER_H__
#define __GST_SUNAUDIO_MIXER_H__
#include "gstsunaudiomixerctrl.h"
G_BEGIN_DECLS
#define GST_SUNAUDIO_MIXER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SUNAUDIO_MIXER,GstSunAudioMixer))
#define GST_SUNAUDIO_MIXER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SUNAUDIO_MIXER,GstSunAudioMixerClass))
#define GST_IS_SUNAUDIO_MIXER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SUNAUDIO_MIXER))
#define GST_IS_SUNAUDIO_MIXER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SUNAUDIO_MIXER))
#define GST_TYPE_SUNAUDIO_MIXER (gst_sunaudiomixer_get_type())
typedef struct _GstSunAudioMixer GstSunAudioMixer;
typedef struct _GstSunAudioMixerClass GstSunAudioMixerClass;
struct _GstSunAudioMixer {
GstElement parent;
GstSunAudioMixerCtrl *mixer;
};
struct _GstSunAudioMixerClass {
GstElementClass parent;
};
GType gst_sunaudiomixer_get_type (void);
G_END_DECLS
#endif /* __GST_SUNAUDIO_MIXER_H__ */

585
sys/sunaudio/gstsunaudiomixerctrl.c
View file

@ -1,585 +0,0 @@
/*
* GStreamer - SunAudio mixer interface element
* Copyright (C) 2005,2006,2008,2009 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Copyright (C) 2008 Sun Microsystems, Inc.,
* Jan Schmidt <jan.schmidt@sun.com>
* Copyright (C) 2009 Sun Microsystems, Inc.,
* Garrett D'Amore <garrett.damore@sun.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, 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>
#include <sys/audio.h>
#include <sys/mixer.h>
#include <gst/gst-i18n-plugin.h>
#include "gstsunaudiomixerctrl.h"
#include "gstsunaudiomixertrack.h"
#include "gstsunaudiomixeroptions.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
static gboolean
gst_sunaudiomixer_ctrl_open (GstSunAudioMixerCtrl * mixer)
{
int fd;
/* First try to open non-blocking */
fd = open (mixer->device, O_RDWR | O_NONBLOCK);
if (fd >= 0) {
close (fd);
fd = open (mixer->device, O_WRONLY);
}
if (fd == -1) {
GST_DEBUG_OBJECT (mixer,
"Failed to open mixer device %s, mixing disabled: %s", mixer->device,
strerror (errno));
return FALSE;
}
mixer->mixer_fd = fd;
/* Try to set the multiple open flag if we can, but ignore errors */
ioctl (mixer->mixer_fd, AUDIO_MIXER_MULTIPLE_OPEN);
GST_DEBUG_OBJECT (mixer, "Opened mixer device %s", mixer->device);
return TRUE;
}
void
gst_sunaudiomixer_ctrl_build_list (GstSunAudioMixerCtrl * mixer)
{
GstMixerTrack *track;
GstMixerOptions *options;
struct audio_info audioinfo;
/*
* Do not continue appending the same 3 static tracks onto the list
*/
if (mixer->tracklist == NULL) {
g_return_if_fail (mixer->mixer_fd != -1);
/* query available ports */
if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
g_warning ("Error getting audio device volume");
return;
}
/* Output & should be MASTER when it's the only one. */
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_OUTPUT);
mixer->tracklist = g_list_append (mixer->tracklist, track);
/* Input */
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_RECORD);
mixer->tracklist = g_list_append (mixer->tracklist, track);
/* Monitor */
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_MONITOR);
mixer->tracklist = g_list_append (mixer->tracklist, track);
if (audioinfo.play.avail_ports & AUDIO_SPEAKER) {
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_SPEAKER);
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
if (audioinfo.play.avail_ports & AUDIO_HEADPHONE) {
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_HP);
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
if (audioinfo.play.avail_ports & AUDIO_LINE_OUT) {
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_LINEOUT);
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
if (audioinfo.play.avail_ports & AUDIO_SPDIF_OUT) {
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_SPDIFOUT);
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
if (audioinfo.play.avail_ports & AUDIO_AUX1_OUT) {
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_AUX1OUT);
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
if (audioinfo.play.avail_ports & AUDIO_AUX2_OUT) {
track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_AUX2OUT);
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
if (audioinfo.record.avail_ports != AUDIO_NONE) {
options =
gst_sunaudiomixer_options_new (mixer, GST_SUNAUDIO_TRACK_RECSRC);
mixer->tracklist = g_list_append (mixer->tracklist, options);
}
}
}
GstSunAudioMixerCtrl *
gst_sunaudiomixer_ctrl_new (const char *device)
{
GstSunAudioMixerCtrl *ret = NULL;
g_return_val_if_fail (device != NULL, NULL);
ret = g_new0 (GstSunAudioMixerCtrl, 1);
ret->device = g_strdup (device);
ret->mixer_fd = -1;
ret->tracklist = NULL;
if (!gst_sunaudiomixer_ctrl_open (ret))
goto error;
return ret;
error:
if (ret)
gst_sunaudiomixer_ctrl_free (ret);
return NULL;
}
void
gst_sunaudiomixer_ctrl_free (GstSunAudioMixerCtrl * mixer)
{
g_return_if_fail (mixer != NULL);
if (mixer->device) {
g_free (mixer->device);
mixer->device = 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);
}
GstMixerFlags
gst_sunaudiomixer_ctrl_get_mixer_flags (GstSunAudioMixerCtrl * mixer)
{
return GST_MIXER_FLAG_HAS_WHITELIST | GST_MIXER_FLAG_GROUPING;
}
const GList *
gst_sunaudiomixer_ctrl_list_tracks (GstSunAudioMixerCtrl * mixer)
{
gst_sunaudiomixer_ctrl_build_list (mixer);
return (const GList *) mixer->tracklist;
}
void
gst_sunaudiomixer_ctrl_get_volume (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track, gint * volumes)
{
gint gain, balance;
float ratio;
struct audio_info audioinfo;
GstSunAudioMixerTrack *sunaudiotrack;
g_return_if_fail (GST_IS_SUNAUDIO_MIXER_TRACK (track));
sunaudiotrack = GST_SUNAUDIO_MIXER_TRACK (track);
g_return_if_fail (mixer->mixer_fd != -1);
if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
g_warning ("Error getting audio device volume");
return;
}
balance = AUDIO_MID_BALANCE;
gain = 0;
switch (sunaudiotrack->track_num) {
case GST_SUNAUDIO_TRACK_OUTPUT:
gain = (int) audioinfo.play.gain;
balance = audioinfo.play.balance;
break;
case GST_SUNAUDIO_TRACK_RECORD:
gain = (int) audioinfo.record.gain;
balance = audioinfo.record.balance;
break;
case GST_SUNAUDIO_TRACK_MONITOR:
gain = (int) audioinfo.monitor_gain;
balance = audioinfo.record.balance;
break;
case GST_SUNAUDIO_TRACK_SPEAKER:
if (audioinfo.play.port & AUDIO_SPEAKER)
gain = AUDIO_MAX_GAIN;
break;
case GST_SUNAUDIO_TRACK_HP:
if (audioinfo.play.port & AUDIO_HEADPHONE)
gain = AUDIO_MAX_GAIN;
break;
case GST_SUNAUDIO_TRACK_LINEOUT:
if (audioinfo.play.port & AUDIO_LINE_OUT)
gain = AUDIO_MAX_GAIN;
break;
case GST_SUNAUDIO_TRACK_SPDIFOUT:
if (audioinfo.play.port & AUDIO_SPDIF_OUT)
gain = AUDIO_MAX_GAIN;
break;
case GST_SUNAUDIO_TRACK_AUX1OUT:
if (audioinfo.play.port & AUDIO_AUX1_OUT)
gain = AUDIO_MAX_GAIN;
break;
case GST_SUNAUDIO_TRACK_AUX2OUT:
if (audioinfo.play.port & AUDIO_AUX2_OUT)
gain = AUDIO_MAX_GAIN;
break;
default:
break;
}
switch (track->num_channels) {
case 2:
if (balance == AUDIO_MID_BALANCE) {
volumes[0] = gain;
volumes[1] = gain;
} else if (balance < AUDIO_MID_BALANCE) {
volumes[0] = gain;
ratio = 1 - (float) (AUDIO_MID_BALANCE - balance) /
(float) AUDIO_MID_BALANCE;
volumes[1] = (int) ((float) gain * ratio + 0.5);
} else {
volumes[1] = gain;
ratio = 1 - (float) (balance - AUDIO_MID_BALANCE) /
(float) AUDIO_MID_BALANCE;
volumes[0] = (int) ((float) gain * ratio + 0.5);
}
break;
case 1:
volumes[0] = gain;
break;
}
/* Likewise reset MUTE */
if ((sunaudiotrack->track_num == GST_SUNAUDIO_TRACK_OUTPUT
&& audioinfo.output_muted == 1)
|| (sunaudiotrack->track_num != GST_SUNAUDIO_TRACK_OUTPUT && gain == 0)) {
/*
* If MUTE is set, then gain is always 0, so don't bother
* resetting our internal value.
*/
track->flags |= GST_MIXER_TRACK_MUTE;
} else {
sunaudiotrack->gain = gain;
sunaudiotrack->balance = balance;
track->flags &= ~GST_MIXER_TRACK_MUTE;
}
}
void
gst_sunaudiomixer_ctrl_set_volume (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track, gint * volumes)
{
gint gain;
gint balance;
gint l_real_gain;
gint r_real_gain;
float ratio;
struct audio_info audioinfo;
GstSunAudioMixerTrack *sunaudiotrack = GST_SUNAUDIO_MIXER_TRACK (track);
l_real_gain = volumes[0];
r_real_gain = volumes[1];
if (l_real_gain == r_real_gain) {
gain = l_real_gain;
balance = AUDIO_MID_BALANCE;
} else if (l_real_gain < r_real_gain) {
gain = r_real_gain;
ratio = (float) l_real_gain / (float) r_real_gain;
balance =
AUDIO_RIGHT_BALANCE - (int) (ratio * (float) AUDIO_MID_BALANCE + 0.5);
} else {
gain = l_real_gain;
ratio = (float) r_real_gain / (float) l_real_gain;
balance =
AUDIO_LEFT_BALANCE + (int) (ratio * (float) AUDIO_MID_BALANCE + 0.5);
}
sunaudiotrack->gain = gain;
sunaudiotrack->balance = balance;
if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_MUTE)) {
if (sunaudiotrack->track_num == GST_SUNAUDIO_TRACK_OUTPUT) {
return;
} else if (gain == 0) {
return;
} else {
/*
* If the volume is set to a non-zero value for LINE_IN
* or MONITOR, then unset MUTE.
*/
track->flags &= ~GST_MIXER_TRACK_MUTE;
}
}
/* Set the volume */
AUDIO_INITINFO (&audioinfo);
switch (sunaudiotrack->track_num) {
case GST_SUNAUDIO_TRACK_OUTPUT:
audioinfo.play.gain = gain;
audioinfo.play.balance = balance;
break;
case GST_SUNAUDIO_TRACK_RECORD:
audioinfo.record.gain = gain;
audioinfo.record.balance = balance;
break;
case GST_SUNAUDIO_TRACK_MONITOR:
audioinfo.monitor_gain = gain;
audioinfo.record.balance = balance;
break;
default:
break;
}
g_return_if_fail (mixer->mixer_fd != -1);
if (ioctl (mixer->mixer_fd, AUDIO_SETINFO, &audioinfo) < 0) {
g_warning ("Error setting audio device volume");
return;
}
}
void
gst_sunaudiomixer_ctrl_set_mute (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track, gboolean mute)
{
struct audio_info audioinfo;
struct audio_info oldinfo;
GstSunAudioMixerTrack *sunaudiotrack = GST_SUNAUDIO_MIXER_TRACK (track);
gint volume, balance;
AUDIO_INITINFO (&audioinfo);
if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &oldinfo) < 0) {
g_warning ("Error getting audio device volume");
return;
}
if (mute) {
volume = 0;
track->flags |= GST_MIXER_TRACK_MUTE;
} else {
volume = sunaudiotrack->gain;
track->flags &= ~GST_MIXER_TRACK_MUTE;
}
balance = sunaudiotrack->balance;
switch (sunaudiotrack->track_num) {
case GST_SUNAUDIO_TRACK_OUTPUT:
if (mute)
audioinfo.output_muted = 1;
else
audioinfo.output_muted = 0;
audioinfo.play.gain = volume;
audioinfo.play.balance = balance;
break;
case GST_SUNAUDIO_TRACK_RECORD:
audioinfo.record.gain = volume;
audioinfo.record.balance = balance;
break;
case GST_SUNAUDIO_TRACK_MONITOR:
audioinfo.monitor_gain = volume;
audioinfo.record.balance = balance;
break;
case GST_SUNAUDIO_TRACK_SPEAKER:
if (mute) {
audioinfo.play.port = oldinfo.play.port & ~AUDIO_SPEAKER;
} else {
audioinfo.play.port = oldinfo.play.port | AUDIO_SPEAKER;
}
break;
case GST_SUNAUDIO_TRACK_HP:
if (mute) {
audioinfo.play.port = oldinfo.play.port & ~AUDIO_HEADPHONE;
} else {
audioinfo.play.port = oldinfo.play.port | AUDIO_HEADPHONE;
}
break;
case GST_SUNAUDIO_TRACK_LINEOUT:
if (mute) {
audioinfo.play.port = oldinfo.play.port & ~AUDIO_LINE_OUT;
} else {
audioinfo.play.port = oldinfo.play.port | AUDIO_LINE_OUT;
}
break;
case GST_SUNAUDIO_TRACK_SPDIFOUT:
if (mute) {
audioinfo.play.port = oldinfo.play.port & ~AUDIO_SPDIF_OUT;
} else {
audioinfo.play.port = oldinfo.play.port | AUDIO_SPDIF_OUT;
}
break;
case GST_SUNAUDIO_TRACK_AUX1OUT:
if (mute) {
audioinfo.play.port = oldinfo.play.port & ~AUDIO_AUX1_OUT;
} else {
audioinfo.play.port = oldinfo.play.port | AUDIO_AUX1_OUT;
}
break;
case GST_SUNAUDIO_TRACK_AUX2OUT:
if (mute) {
audioinfo.play.port = oldinfo.play.port & ~AUDIO_AUX2_OUT;
} else {
audioinfo.play.port = oldinfo.play.port | AUDIO_AUX2_OUT;
}
break;
default:
break;
}
if (audioinfo.play.port != ((unsigned) ~0)) {
/* mask off ports we can't modify. Hack for broken drivers where mod_ports == 0 */
if (oldinfo.play.mod_ports != 0) {
audioinfo.play.port &= oldinfo.play.mod_ports;
/* and add in any that are forced to be on */
audioinfo.play.port |= (oldinfo.play.port & ~oldinfo.play.mod_ports);
}
}
g_return_if_fail (mixer->mixer_fd != -1);
if (audioinfo.play.port != (guint) (-1) &&
audioinfo.play.port != oldinfo.play.port)
GST_LOG_OBJECT (mixer, "Changing play port mask to 0x%08x",
audioinfo.play.port);
if (ioctl (mixer->mixer_fd, AUDIO_SETINFO, &audioinfo) < 0) {
g_warning ("Error setting audio settings");
return;
}
}
void
gst_sunaudiomixer_ctrl_set_record (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track, gboolean record)
{
}
void
gst_sunaudiomixer_ctrl_set_option (GstSunAudioMixerCtrl * mixer,
GstMixerOptions * options, gchar * value)
{
struct audio_info audioinfo;
GstMixerTrack *track;
GstSunAudioMixerOptions *opts;
GQuark q;
int i;
g_return_if_fail (mixer != NULL);
g_return_if_fail (mixer->mixer_fd != -1);
g_return_if_fail (value != NULL);
g_return_if_fail (GST_IS_SUNAUDIO_MIXER_OPTIONS (options));
track = GST_MIXER_TRACK (options);
opts = GST_SUNAUDIO_MIXER_OPTIONS (options);
if (opts->track_num != GST_SUNAUDIO_TRACK_RECSRC) {
g_warning ("set_option not supported on track %s", track->label);
return;
}
q = g_quark_try_string (value);
if (q == 0) {
g_warning ("unknown option '%s'", value);
return;
}
for (i = 0; i < 8; i++) {
if (opts->names[i] == q) {
break;
}
}
if (((1 << (i)) & opts->avail) == 0) {
g_warning ("Record port %s not available", g_quark_to_string (q));
return;
}
AUDIO_INITINFO (&audioinfo);
audioinfo.record.port = (1 << (i));
if (ioctl (mixer->mixer_fd, AUDIO_SETINFO, &audioinfo) < 0) {
g_warning ("Error setting audio record port");
}
}
const gchar *
gst_sunaudiomixer_ctrl_get_option (GstSunAudioMixerCtrl * mixer,
GstMixerOptions * options)
{
GstMixerTrack *track;
GstSunAudioMixerOptions *opts;
struct audio_info audioinfo;
int i;
g_return_val_if_fail (mixer != NULL, NULL);
g_return_val_if_fail (mixer->fd != -1, NULL);
g_return_val_if_fail (GST_IS_SUNAUDIO_MIXER_OPTIONS (options), NULL);
track = GST_MIXER_TRACK (options);
opts = GST_SUNAUDIO_MIXER_OPTIONS (options);
g_return_val_if_fail (opts->track_num == GST_SUNAUDIO_TRACK_RECSRC, NULL);
if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
g_warning ("Error getting audio device settings");
return (NULL);
}
for (i = 0; i < 8; i++) {
if ((1 << i) == audioinfo.record.port) {
const gchar *s = g_quark_to_string (opts->names[i]);
GST_DEBUG_OBJECT (mixer, "Getting value for option %d: %s",
opts->track_num, s);
return (s);
}
}
GST_DEBUG_OBJECT (mixer, "Unable to get value for option %d",
opts->track_num);
g_warning ("Record port value %d seems illegal", audioinfo.record.port);
return (NULL);
}

189
sys/sunaudio/gstsunaudiomixerctrl.h
View file

@ -1,189 +0,0 @@
/*
* GStreamer - SunAudio mixer interface element.
* Copyright (C) 2005,2006,2009 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Copyright (C) 2009 Sun Microsystems, Inc.,
* Garrett D'Amore <garrett.damore@sun.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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#ifndef __GST_SUNAUDIO_MIXER_CTRL_H
#define __GST_SUNAUDIO_MIXER_CTRL_H
#include <sys/audioio.h>
#include <gst/gst.h>
#include <gst/interfaces/mixer.h>
G_BEGIN_DECLS
#define GST_SUNAUDIO_MIXER_CTRL(obj) ((GstSunAudioMixerCtrl*)(obj))
typedef struct _GstSunAudioMixerCtrl GstSunAudioMixerCtrl;
struct _GstSunAudioMixerCtrl {
GList * tracklist; /* list of available tracks */
gint fd;
gint mixer_fd;
gchar * device;
};
GstSunAudioMixerCtrl* gst_sunaudiomixer_ctrl_new (const gchar *device);
void gst_sunaudiomixer_ctrl_free (GstSunAudioMixerCtrl *mixer);
const GList* gst_sunaudiomixer_ctrl_list_tracks (GstSunAudioMixerCtrl * mixer);
void gst_sunaudiomixer_ctrl_set_volume (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track,
gint * volumes);
void gst_sunaudiomixer_ctrl_get_volume (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track,
gint * volumes);
void gst_sunaudiomixer_ctrl_set_record (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track,
gboolean record);
void gst_sunaudiomixer_ctrl_set_mute (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track,
gboolean mute);
void gst_sunaudiomixer_ctrl_set_option (GstSunAudioMixerCtrl * mixer,
GstMixerOptions * options,
gchar * value);
const gchar * gst_sunaudiomixer_ctrl_get_option (GstSunAudioMixerCtrl * mixer,
GstMixerOptions * options);
GstMixerFlags gst_sunaudiomixer_ctrl_get_mixer_flags (GstSunAudioMixerCtrl *mixer);
#define GST_IMPLEMENT_SUNAUDIO_MIXER_CTRL_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_sunaudiomixer_ctrl_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_sunaudiomixer_ctrl_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_sunaudiomixer_ctrl_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_sunaudiomixer_ctrl_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_sunaudiomixer_ctrl_set_mute (this->mixer, track, mute); \
} \
\
static const gchar * \
interface_as_function ## _get_option (GstMixer * mixer, GstMixerOptions * opts) \
{ \
Type *this = (Type*) mixer; \
\
g_return_val_if_fail (this != NULL, NULL); \
g_return_val_if_fail (this->mixer != NULL, NULL); \
\
return gst_sunaudiomixer_ctrl_get_option (this->mixer, opts); \
} \
\
static void \
interface_as_function ## _set_option (GstMixer * mixer, GstMixerOptions * opts, \
gchar * value) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_sunaudiomixer_ctrl_set_option (this->mixer, opts, value); \
} \
\
static GstMixerFlags \
interface_as_function ## _get_mixer_flags (GstMixer * mixer) \
{ \
Type *this = (Type*) mixer; \
\
g_return_val_if_fail (this != NULL, GST_MIXER_FLAG_NONE); \
g_return_val_if_fail (this->mixer != NULL, GST_MIXER_FLAG_NONE); \
\
return gst_sunaudiomixer_ctrl_get_mixer_flags (this->mixer); \
} \
\
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; \
iface->get_option = interface_as_function ## _get_option; \
iface->set_option = interface_as_function ## _set_option; \
iface->get_mixer_flags = interface_as_function ## _get_mixer_flags; \
}
G_END_DECLS
#endif

157
sys/sunaudio/gstsunaudiomixeroptions.c
View file

@ -1,157 +0,0 @@
/*
* GStreamer SunAudio mixer track implementation
* Copyright (C) 2009 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Garrett D'Amore <garrett.damore@sun.com>
*
* gstsunaudiomixeroptions.c: Sun Audio mixer options object
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, 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>
#include <sys/audio.h>
#include <sys/mixer.h>
#include <gst/gst-i18n-plugin.h>
#include "gstsunaudiomixeroptions.h"
#include "gstsunaudiomixertrack.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
static void gst_sunaudiomixer_options_init (GstSunAudioMixerOptions * sun_opts);
static void gst_sunaudiomixer_options_class_init (gpointer g_class,
gpointer class_data);
static GstMixerOptionsClass *parent_class = NULL;
GType
gst_sunaudiomixer_options_get_type (void)
{
static GType opts_type = 0;
if (!opts_type) {
static const GTypeInfo opts_info = {
sizeof (GstSunAudioMixerOptionsClass),
NULL,
NULL,
gst_sunaudiomixer_options_class_init,
NULL,
NULL,
sizeof (GstSunAudioMixerOptions),
0,
(GInstanceInitFunc) gst_sunaudiomixer_options_init,
};
opts_type =
g_type_register_static (GST_TYPE_MIXER_OPTIONS,
"GstSunAudioMixerOptions", &opts_info, 0);
}
return opts_type;
}
static void
gst_sunaudiomixer_options_class_init (gpointer g_class, gpointer class_data)
{
parent_class = g_type_class_peek_parent (g_class);
}
static void
gst_sunaudiomixer_options_init (GstSunAudioMixerOptions * sun_opts)
{
}
GstMixerOptions *
gst_sunaudiomixer_options_new (GstSunAudioMixerCtrl * mixer, gint track_num)
{
GstMixerOptions *opts;
GstSunAudioMixerOptions *sun_opts;
GstMixerTrack *track;
const gchar *label;
gint i;
struct audio_info audioinfo;
if ((mixer == NULL) || (mixer->mixer_fd == -1)) {
g_warning ("mixer not initialized");
return NULL;
}
if (track_num != GST_SUNAUDIO_TRACK_RECSRC) {
g_warning ("invalid options track");
return (NULL);
}
label = N_("Record Source");
opts = g_object_new (GST_TYPE_SUNAUDIO_MIXER_OPTIONS,
"untranslated-label", label, NULL);
sun_opts = GST_SUNAUDIO_MIXER_OPTIONS (opts);
track = GST_MIXER_TRACK (opts);
GST_DEBUG_OBJECT (opts, "New mixer options, track %d: %s",
track_num, GST_STR_NULL (label));
/* save off names for the record sources */
sun_opts->names[0] = g_quark_from_string (_("Microphone"));
sun_opts->names[1] = g_quark_from_string (_("Line In"));
sun_opts->names[2] = g_quark_from_string (_("Internal CD"));
sun_opts->names[3] = g_quark_from_string (_("SPDIF In"));
sun_opts->names[4] = g_quark_from_string (_("AUX 1 In"));
sun_opts->names[5] = g_quark_from_string (_("AUX 2 In"));
sun_opts->names[6] = g_quark_from_string (_("Codec Loopback"));
sun_opts->names[7] = g_quark_from_string (_("SunVTS Loopback"));
/* set basic information */
track->label = g_strdup (_(label));
track->num_channels = 0;
track->min_volume = 0;
track->max_volume = 0;
track->flags =
GST_MIXER_TRACK_INPUT | GST_MIXER_TRACK_WHITELIST |
GST_MIXER_TRACK_NO_RECORD;
if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
g_warning ("Error getting audio device settings");
g_object_unref (G_OBJECT (sun_opts));
return NULL;
}
sun_opts->avail = audioinfo.record.avail_ports;
sun_opts->track_num = track_num;
for (i = 0; i < 8; i++) {
if ((1 << i) & audioinfo.record.avail_ports) {
const char *s = g_quark_to_string (sun_opts->names[i]);
opts->values = g_list_append (opts->values, g_strdup (s));
GST_DEBUG_OBJECT (opts, "option for track %d: %s",
track_num, GST_STR_NULL (s));
}
}
return opts;
}

65
sys/sunaudio/gstsunaudiomixeroptions.h
View file

@ -1,65 +0,0 @@
/*
* GStreamer SunAudio mixer track implementation
* Copyright (C) 2009 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Garrett D'Amore <garrett.damore@sun.com>
*
* gstsunaudiomixeroptions.h: Sun Audio mixer options object
*
* 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., 51 Franklin St, Fifth Floor,
*/
#ifndef __GST_SUNAUDIO_MIXER_OPTIONS_H__
#define __GST_SUNAUDIO_MIXER_OPTIONS_H__
#include "gstsunaudiomixer.h"
#include <gst/interfaces/mixeroptions.h>
G_BEGIN_DECLS
#define GST_SUNAUDIO_MIXER_OPTIONS(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SUNAUDIO_MIXER_OPTIONS, GstSunAudioMixerOptions))
#define GST_SUNAUDIO_MIXER_OPTIONS_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SUNAUDIO_MIXER_OPTIONS, GstSunAudioMixerOptionsClass))
#define GST_IS_SUNAUDIO_MIXER_OPTIONS(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SUNAUDIO_MIXER_OPTIONS))
#define GST_IS_SUNAUDIO_MIXER_OPTIONS_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SUNAUDIO_MIXER_OPTIONS))
#define GST_TYPE_SUNAUDIO_MIXER_OPTIONS (gst_sunaudiomixer_options_get_type())
typedef struct _GstSunAudioMixerOptions GstSunAudioMixerOptions;
typedef struct _GstSunAudioMixerOptionsClass GstSunAudioMixerOptionsClass;
struct _GstSunAudioMixerOptions {
GstMixerOptions parent;
gint track_num;
GQuark names[8]; /* only 8 possible */
gint avail; /* mask of avail */
};
struct _GstSunAudioMixerOptionsClass {
GstMixerOptionsClass parent;
};
GType gst_sunaudiomixer_options_get_type (void);
GstMixerOptions *gst_sunaudiomixer_options_new (GstSunAudioMixerCtrl *mixer, gint track_num);
G_END_DECLS
#endif /* __GST_SUNAUDIO_MIXER_OPTIONS_H__ */

158
sys/sunaudio/gstsunaudiomixertrack.c
View file

@ -1,158 +0,0 @@
/*
* GStreamer
* Copyright (C) 2005,2008, 2009 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Copyright (C) 2009 Sun Microsystems, Inc.,
* Garrett D'Amore <garrett.damore@sun.com>
*
* gstsunaudiomixer.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, 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>
#include <sys/audioio.h>
#include <gst/gst-i18n-plugin.h>
#include "gstsunaudiomixertrack.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
#define MASK_BIT_IS_SET(mask, bit) \
(mask & (1 << bit))
G_DEFINE_TYPE (GstSunAudioMixerTrack, gst_sunaudiomixer_track,
GST_TYPE_MIXER_TRACK);
static void
gst_sunaudiomixer_track_class_init (GstSunAudioMixerTrackClass * klass)
{
/* nop */
}
static void
gst_sunaudiomixer_track_init (GstSunAudioMixerTrack * track)
{
track->gain = 0;
track->balance = AUDIO_MID_BALANCE;
track->track_num = 0;
}
GstMixerTrack *
gst_sunaudiomixer_track_new (GstSunAudioTrackType track_num)
{
const gchar *labels[] = { N_("Volume"),
N_("Gain"),
N_("Monitor"),
N_("Built-in Speaker"),
N_("Headphone"),
N_("Line Out"),
N_("SPDIF Out"),
N_("AUX 1 Out"),
N_("AUX 2 Out"),
};
GstSunAudioMixerTrack *sunaudiotrack;
GstMixerTrack *track;
const gchar *untranslated_label;
if ((guint) track_num < G_N_ELEMENTS (labels))
untranslated_label = labels[track_num];
else
untranslated_label = NULL;
sunaudiotrack = g_object_new (GST_TYPE_SUNAUDIO_MIXER_TRACK,
"untranslated-label", untranslated_label, NULL);
GST_DEBUG_OBJECT (sunaudiotrack, "Creating new mixer track of type %d: %s",
track_num, GST_STR_NULL (untranslated_label));
switch (track_num) {
case GST_SUNAUDIO_TRACK_OUTPUT:
/* these are sliders */
track = GST_MIXER_TRACK (sunaudiotrack);
track->label = g_strdup (_(untranslated_label));
track->num_channels = 2;
track->flags = GST_MIXER_TRACK_OUTPUT | GST_MIXER_TRACK_WHITELIST |
GST_MIXER_TRACK_MASTER;
track->min_volume = 0;
track->max_volume = 255;
sunaudiotrack->track_num = track_num;
sunaudiotrack->gain = (0 & 0xff);
sunaudiotrack->balance = AUDIO_MID_BALANCE;
break;
case GST_SUNAUDIO_TRACK_RECORD:
/* these are sliders */
track = GST_MIXER_TRACK (sunaudiotrack);
track->label = g_strdup (_(untranslated_label));
track->num_channels = 2;
track->flags = GST_MIXER_TRACK_INPUT | GST_MIXER_TRACK_NO_RECORD |
GST_MIXER_TRACK_WHITELIST;
track->min_volume = 0;
track->max_volume = 255;
sunaudiotrack->track_num = track_num;
sunaudiotrack->gain = (0 & 0xff);
sunaudiotrack->balance = AUDIO_MID_BALANCE;
break;
case GST_SUNAUDIO_TRACK_MONITOR:
/* these are sliders */
track = GST_MIXER_TRACK (sunaudiotrack);
track->label = g_strdup (_(untranslated_label));
track->num_channels = 2;
track->flags = GST_MIXER_TRACK_INPUT | GST_MIXER_TRACK_NO_RECORD;
track->min_volume = 0;
track->max_volume = 255;
sunaudiotrack->track_num = track_num;
sunaudiotrack->gain = (0 & 0xff);
sunaudiotrack->balance = AUDIO_MID_BALANCE;
break;
case GST_SUNAUDIO_TRACK_SPEAKER:
case GST_SUNAUDIO_TRACK_HP:
case GST_SUNAUDIO_TRACK_LINEOUT:
case GST_SUNAUDIO_TRACK_SPDIFOUT:
case GST_SUNAUDIO_TRACK_AUX1OUT:
case GST_SUNAUDIO_TRACK_AUX2OUT:
/* these are switches */
track = GST_MIXER_TRACK (sunaudiotrack);
track->label = g_strdup (_(untranslated_label));
track->num_channels = 0;
track->flags = GST_MIXER_TRACK_OUTPUT | GST_MIXER_TRACK_WHITELIST;
track->min_volume = 0;
track->max_volume = 255;
sunaudiotrack->track_num = track_num;
sunaudiotrack->gain = (0 & 0xff);
sunaudiotrack->balance = AUDIO_MID_BALANCE;
break;
default:
g_warning ("Unknown sun audio track num %d", track_num);
track = NULL;
}
return track;
}

78
sys/sunaudio/gstsunaudiomixertrack.h
View file

@ -1,78 +0,0 @@
/*
* GStreamer SunAudio mixer track implementation
* Copyright (C) 2005,2006,2009 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Copyright (C) 2009 Sun Microsystems, Inc.,
* Garrett D'Amore <garrett.damore@sun.com>
*
* gstsunaudiomixertrack.h: SunAudio 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_SUNAUDIO_MIXER_TRACK_H__
#define __GST_SUNAUDIO_MIXER_TRACK_H__
#include <gst/gst.h>
#include <gst/interfaces/mixer.h>
G_BEGIN_DECLS
typedef enum
{
GST_SUNAUDIO_TRACK_OUTPUT = 0,
GST_SUNAUDIO_TRACK_RECORD,
GST_SUNAUDIO_TRACK_MONITOR,
GST_SUNAUDIO_TRACK_SPEAKER,
GST_SUNAUDIO_TRACK_HP,
GST_SUNAUDIO_TRACK_LINEOUT,
GST_SUNAUDIO_TRACK_SPDIFOUT,
GST_SUNAUDIO_TRACK_AUX1OUT,
GST_SUNAUDIO_TRACK_AUX2OUT,
GST_SUNAUDIO_TRACK_RECSRC
} GstSunAudioTrackType;
#define GST_TYPE_SUNAUDIO_MIXER_TRACK \
(gst_sunaudiomixer_track_get_type ())
#define GST_SUNAUDIO_MIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_SUNAUDIO_MIXER_TRACK, \
GstSunAudioMixerTrack))
#define GST_SUNAUDIO_MIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_SUNAUDIO_MIXER_TRACK, \
GstSunAudioMixerTrackClass))
#define GST_IS_SUNAUDIO_MIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_SUNAUDIO_MIXER_TRACK))
#define GST_IS_SUNAUDIO_MIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_SUNAUDIO_MIXER_TRACK))
typedef struct _GstSunAudioMixerTrack {
GstMixerTrack parent;
gint gain;
gint balance;
GstSunAudioTrackType track_num;
} GstSunAudioMixerTrack;
typedef struct _GstSunAudioMixerTrackClass {
GstMixerTrackClass parent;
} GstSunAudioMixerTrackClass;
GType gst_sunaudiomixer_track_get_type (void);
GstMixerTrack* gst_sunaudiomixer_track_new (GstSunAudioTrackType track_num);
G_END_DECLS
#endif /* __GST_SUNAUDIO_MIXER_TRACK_H__ */

650
sys/sunaudio/gstsunaudiosink.c
View file

@ -1,650 +0,0 @@
/*
* GStreamer - SunAudio sink
* Copyright (C) 2004 David A. Schleef <ds@schleef.org>
* Copyright (C) 2005,2006 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Copyright (C) 2006 Jan Schmidt <thaytan@mad.scientist.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-sunaudiosink
*
* sunaudiosink is an audio sink designed to work with the Sun Audio
* interface available in Solaris.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 audiotestsrc volume=0.5 ! sunaudiosink
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <fcntl.h>
#include <string.h>
#include <stropts.h>
#include <unistd.h>
#include <sys/mman.h>
#include "gstsunaudiosink.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
static void gst_sunaudiosink_base_init (gpointer g_class);
static void gst_sunaudiosink_class_init (GstSunAudioSinkClass * klass);
static void gst_sunaudiosink_init (GstSunAudioSink * filter);
static void gst_sunaudiosink_dispose (GObject * object);
static void gst_sunaudiosink_finalize (GObject * object);
static void gst_sunaudiosink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_sunaudiosink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_sunaudiosink_getcaps (GstBaseSink * bsink);
static gboolean gst_sunaudiosink_open (GstAudioSink * asink);
static gboolean gst_sunaudiosink_close (GstAudioSink * asink);
static gboolean gst_sunaudiosink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_sunaudiosink_unprepare (GstAudioSink * asink);
static guint gst_sunaudiosink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_sunaudiosink_delay (GstAudioSink * asink);
static void gst_sunaudiosink_reset (GstAudioSink * asink);
#define DEFAULT_DEVICE "/dev/audio"
enum
{
PROP_0,
PROP_DEVICE,
};
static GstStaticPadTemplate gst_sunaudiosink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16, "
/* [5510,48000] seems to be a Solaris limit */
"rate = (int) [ 5510, 48000 ], " "channels = (int) [ 1, 2 ]")
);
static GstElementClass *parent_class = NULL;
GType
gst_sunaudiosink_get_type (void)
{
static GType plugin_type = 0;
if (!plugin_type) {
static const GTypeInfo plugin_info = {
sizeof (GstSunAudioSinkClass),
gst_sunaudiosink_base_init,
NULL,
(GClassInitFunc) gst_sunaudiosink_class_init,
NULL,
NULL,
sizeof (GstSunAudioSink),
0,
(GInstanceInitFunc) gst_sunaudiosink_init,
};
plugin_type = g_type_register_static (GST_TYPE_AUDIO_SINK,
"GstSunAudioSink", &plugin_info, 0);
}
return plugin_type;
}
static void
gst_sunaudiosink_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_sunaudiosink_finalize (GObject * object)
{
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (object);
g_mutex_free (sunaudiosink->write_mutex);
g_cond_free (sunaudiosink->sleep_cond);
g_free (sunaudiosink->device);
if (sunaudiosink->fd != -1) {
close (sunaudiosink->fd);
sunaudiosink->fd = -1;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_sunaudiosink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_static_pad_template (element_class,
&gst_sunaudiosink_factory);
gst_element_class_set_static_metadata (element_class, "Sun Audio Sink",
"Sink/Audio", "Audio sink for Sun Audio devices",
"David A. Schleef <ds@schleef.org>, "
"Brian Cameron <brian.cameron@sun.com>");
}
static void
gst_sunaudiosink_class_init (GstSunAudioSinkClass * 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_peek_parent (klass);
gobject_class->dispose = gst_sunaudiosink_dispose;
gobject_class->finalize = gst_sunaudiosink_finalize;
gobject_class->set_property = gst_sunaudiosink_set_property;
gobject_class->get_property = gst_sunaudiosink_get_property;
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_sunaudiosink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_sunaudiosink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_sunaudiosink_close);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_sunaudiosink_prepare);
gstaudiosink_class->unprepare =
GST_DEBUG_FUNCPTR (gst_sunaudiosink_unprepare);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_sunaudiosink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_sunaudiosink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_sunaudiosink_reset);
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device", "Audio Device (/dev/audio)",
DEFAULT_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_sunaudiosink_init (GstSunAudioSink * sunaudiosink)
{
const char *audiodev;
GST_DEBUG_OBJECT (sunaudiosink, "initializing sunaudiosink");
sunaudiosink->fd = -1;
audiodev = g_getenv ("AUDIODEV");
if (audiodev == NULL)
audiodev = DEFAULT_DEVICE;
sunaudiosink->device = g_strdup (audiodev);
/* mutex and gcond used to control the write method */
sunaudiosink->write_mutex = g_mutex_new ();
sunaudiosink->sleep_cond = g_cond_new ();
}
static void
gst_sunaudiosink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSunAudioSink *sunaudiosink;
sunaudiosink = GST_SUNAUDIO_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
GST_OBJECT_LOCK (sunaudiosink);
g_free (sunaudiosink->device);
sunaudiosink->device = g_strdup (g_value_get_string (value));
GST_OBJECT_UNLOCK (sunaudiosink);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_sunaudiosink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSunAudioSink *sunaudiosink;
sunaudiosink = GST_SUNAUDIO_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
GST_OBJECT_LOCK (sunaudiosink);
g_value_set_string (value, sunaudiosink->device);
GST_OBJECT_UNLOCK (sunaudiosink);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_sunaudiosink_getcaps (GstBaseSink * bsink)
{
GstPadTemplate *pad_template;
GstCaps *caps = NULL;
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (bsink);
GST_DEBUG_OBJECT (sunaudiosink, "getcaps called");
pad_template = gst_static_pad_template_get (&gst_sunaudiosink_factory);
caps = gst_caps_copy (gst_pad_template_get_caps (pad_template));
gst_object_unref (pad_template);
return caps;
}
static gboolean
gst_sunaudiosink_open (GstAudioSink * asink)
{
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
int fd, ret;
/* First try to open non-blocking */
GST_OBJECT_LOCK (sunaudiosink);
fd = open (sunaudiosink->device, O_WRONLY | O_NONBLOCK);
if (fd >= 0) {
close (fd);
fd = open (sunaudiosink->device, O_WRONLY);
}
if (fd == -1) {
GST_OBJECT_UNLOCK (sunaudiosink);
goto open_failed;
}
sunaudiosink->fd = fd;
GST_OBJECT_UNLOCK (sunaudiosink);
ret = ioctl (fd, AUDIO_GETDEV, &sunaudiosink->dev);
if (ret == -1)
goto ioctl_error;
GST_DEBUG_OBJECT (sunaudiosink, "name %s", sunaudiosink->dev.name);
GST_DEBUG_OBJECT (sunaudiosink, "version %s", sunaudiosink->dev.version);
GST_DEBUG_OBJECT (sunaudiosink, "config %s", sunaudiosink->dev.config);
ret = ioctl (fd, AUDIO_GETINFO, &sunaudiosink->info);
if (ret == -1)
goto ioctl_error;
GST_DEBUG_OBJECT (sunaudiosink, "monitor_gain %d",
sunaudiosink->info.monitor_gain);
GST_DEBUG_OBJECT (sunaudiosink, "output_muted %d",
sunaudiosink->info.output_muted);
GST_DEBUG_OBJECT (sunaudiosink, "hw_features %08x",
sunaudiosink->info.hw_features);
GST_DEBUG_OBJECT (sunaudiosink, "sw_features %08x",
sunaudiosink->info.sw_features);
GST_DEBUG_OBJECT (sunaudiosink, "sw_features_enabled %08x",
sunaudiosink->info.sw_features_enabled);
return TRUE;
open_failed:
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, OPEN_WRITE, (NULL),
("can't open connection to Sun Audio device %s", sunaudiosink->device));
return FALSE;
ioctl_error:
close (sunaudiosink->fd);
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
static gboolean
gst_sunaudiosink_close (GstAudioSink * asink)
{
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
if (sunaudiosink->fd != -1) {
close (sunaudiosink->fd);
sunaudiosink->fd = -1;
}
return TRUE;
}
static gboolean
gst_sunaudiosink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
audio_info_t ainfo;
int ret;
int ports;
ret = ioctl (sunaudiosink->fd, AUDIO_GETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
if (spec->width != 16)
return FALSE;
ports = ainfo.play.port;
AUDIO_INITINFO (&ainfo);
ainfo.play.sample_rate = spec->rate;
ainfo.play.channels = spec->channels;
ainfo.play.precision = spec->width;
ainfo.play.encoding = AUDIO_ENCODING_LINEAR;
ainfo.play.port = ports;
/* buffer_time for playback is not implemented in Solaris at the moment,
but at some point in the future, it might be */
ainfo.play.buffer_size =
gst_util_uint64_scale (spec->rate * spec->bytes_per_sample,
spec->buffer_time, GST_SECOND / GST_USECOND);
spec->silence_sample[0] = 0;
spec->silence_sample[1] = 0;
spec->silence_sample[2] = 0;
spec->silence_sample[3] = 0;
ret = ioctl (sunaudiosink->fd, AUDIO_SETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
/* Now read back the info to find out the actual buffer size and set
segtotal */
AUDIO_INITINFO (&ainfo);
ret = ioctl (sunaudiosink->fd, AUDIO_GETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
#if 0
/* We don't actually use the buffer_size from the sound device, because
* it seems it's just bogus sometimes */
sunaudiosink->segtotal = spec->segtotal =
ainfo.play.buffer_size / spec->segsize;
#else
sunaudiosink->segtotal = spec->segtotal;
#endif
sunaudiosink->segtotal_samples =
spec->segtotal * spec->segsize / spec->bytes_per_sample;
sunaudiosink->segs_written = (gint) ainfo.play.eof;
sunaudiosink->samples_written = ainfo.play.samples;
sunaudiosink->bytes_per_sample = spec->bytes_per_sample;
GST_DEBUG_OBJECT (sunaudiosink, "Got device buffer_size of %u",
ainfo.play.buffer_size);
return TRUE;
}
static gboolean
gst_sunaudiosink_unprepare (GstAudioSink * asink)
{
return TRUE;
}
#define LOOP_WHILE_EINTR(v,func) do { (v) = (func); } \
while ((v) == -1 && errno == EINTR);
/* Called with the write_mutex held */
static void
gst_sunaudio_sink_do_delay (GstSunAudioSink * sink)
{
GstBaseAudioSink *ba_sink = GST_BASE_AUDIO_SINK (sink);
GstClockTime total_sleep;
GstClockTime max_sleep;
gint sleep_usecs;
GTimeVal sleep_end;
gint err;
audio_info_t ainfo;
guint diff;
/* This code below ensures that we don't race any further than buffer_time
* ahead of the audio output, by sleeping if the next write call would cause
* us to advance too far in the ring-buffer */
LOOP_WHILE_EINTR (err, ioctl (sink->fd, AUDIO_GETINFO, &ainfo));
if (err < 0)
goto write_error;
/* Compute our offset from the output (copes with overflow) */
diff = (guint) (sink->segs_written) - ainfo.play.eof;
if (diff > sink->segtotal) {
/* This implies that reset did a flush just as the sound device aquired
* some buffers internally, and it causes us to be out of sync with the
* eof measure. This corrects it */
sink->segs_written = ainfo.play.eof;
diff = 0;
}
if (diff + 1 < sink->segtotal)
return; /* no need to sleep at all */
/* Never sleep longer than the initial number of undrained segments in the
device plus one */
total_sleep = 0;
max_sleep = (diff + 1) * (ba_sink->latency_time * GST_USECOND);
/* sleep for a segment period between .eof polls */
sleep_usecs = ba_sink->latency_time;
/* Current time is our reference point */
g_get_current_time (&sleep_end);
/* If the next segment would take us too far along the ring buffer,
* sleep for a bit to free up a slot. If there were a way to find out
* when the eof field actually increments, we could use, but the only
* notification mechanism seems to be SIGPOLL, which we can't use from
* a support library */
while (diff + 1 >= sink->segtotal && total_sleep < max_sleep) {
GST_LOG_OBJECT (sink, "need to block to drain segment(s). "
"Sleeping for %d us", sleep_usecs);
g_time_val_add (&sleep_end, sleep_usecs);
if (g_cond_timed_wait (sink->sleep_cond, sink->write_mutex, &sleep_end)) {
GST_LOG_OBJECT (sink, "Waking up early due to reset");
return; /* Got told to wake up */
}
total_sleep += (sleep_usecs * GST_USECOND);
LOOP_WHILE_EINTR (err, ioctl (sink->fd, AUDIO_GETINFO, &ainfo));
if (err < 0)
goto write_error;
/* Compute our (new) offset from the output (copes with overflow) */
diff = (guint) g_atomic_int_get (&sink->segs_written) - ainfo.play.eof;
}
return;
write_error:
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
("Playback error on device '%s': %s", sink->device, strerror (errno)));
return;
}
static guint
gst_sunaudiosink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstSunAudioSink *sink = GST_SUNAUDIO_SINK (asink);
gint bytes_written, err;
g_mutex_lock (sink->write_mutex);
if (sink->flushing) {
/* Exit immediately if reset tells us to */
g_mutex_unlock (sink->write_mutex);
return length;
}
LOOP_WHILE_EINTR (bytes_written, write (sink->fd, data, length));
if (bytes_written < 0) {
err = bytes_written;
goto write_error;
}
/* Increment our sample counter, for delay calcs */
g_atomic_int_add (&sink->samples_written, length / sink->bytes_per_sample);
/* Don't consider the segment written if we didn't output the whole lot yet */
if (bytes_written < length) {
g_mutex_unlock (sink->write_mutex);
return (guint) bytes_written;
}
/* Write a zero length output to trigger increment of the eof field */
LOOP_WHILE_EINTR (err, write (sink->fd, NULL, 0));
if (err < 0)
goto write_error;
/* Count this extra segment we've written */
sink->segs_written += 1;
/* Now delay so we don't overrun the ring buffer */
gst_sunaudio_sink_do_delay (sink);
g_mutex_unlock (sink->write_mutex);
return length;
write_error:
g_mutex_unlock (sink->write_mutex);
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
("Playback error on device '%s': %s", sink->device, strerror (errno)));
return length; /* Say we wrote the segment to let the ringbuffer exit */
}
/*
* Provide the current number of unplayed samples that have been written
* to the device */
static guint
gst_sunaudiosink_delay (GstAudioSink * asink)
{
GstSunAudioSink *sink = GST_SUNAUDIO_SINK (asink);
audio_info_t ainfo;
gint ret;
guint offset;
ret = ioctl (sink->fd, AUDIO_GETINFO, &ainfo);
if (G_UNLIKELY (ret == -1))
return 0;
offset = (g_atomic_int_get (&sink->samples_written) - ainfo.play.samples);
/* If the offset is larger than the total ringbuffer size, then we asked
between the write call and when samples_written is updated */
if (G_UNLIKELY (offset > sink->segtotal_samples))
return 0;
return offset;
}
static void
gst_sunaudiosink_reset (GstAudioSink * asink)
{
/* Get current values */
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
audio_info_t ainfo;
int ret;
ret = ioctl (sunaudiosink->fd, AUDIO_GETINFO, &ainfo);
if (ret == -1) {
/*
* Should never happen, but if we couldn't getinfo, then no point
* trying to setinfo
*/
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return;
}
/*
* Pause the audio - so audio stops playing immediately rather than
* waiting for the ringbuffer to empty.
*/
ainfo.play.pause = !NULL;
ret = ioctl (sunaudiosink->fd, AUDIO_SETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
}
/* Flush the audio */
ret = ioctl (sunaudiosink->fd, I_FLUSH, FLUSHW);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
}
/* Now, we take the write_mutex and signal to ensure the write thread
* is not busy, and we signal the condition to wake up any sleeper,
* then we flush again in case the write wrote something after we flushed,
* and finally release the lock and unpause */
g_mutex_lock (sunaudiosink->write_mutex);
sunaudiosink->flushing = TRUE;
g_cond_signal (sunaudiosink->sleep_cond);
ret = ioctl (sunaudiosink->fd, I_FLUSH, FLUSHW);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
}
/* unpause the audio */
ainfo.play.pause = NULL;
ret = ioctl (sunaudiosink->fd, AUDIO_SETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
}
/* After flushing the audio device, we need to remeasure the sample count
* and segments written count so we're in sync with the device */
sunaudiosink->segs_written = ainfo.play.eof;
g_atomic_int_set (&sunaudiosink->samples_written, ainfo.play.samples);
sunaudiosink->flushing = FALSE;
g_mutex_unlock (sunaudiosink->write_mutex);
}

78
sys/sunaudio/gstsunaudiosink.h
View file

@ -1,78 +0,0 @@
/*
* GStreamer - SunAudio sink
* Copyright (C) 2004 David A. Schleef <ds@schleef.org>
* Copyright (C) 2005,2006 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
* Copyright (C) 2006 Jan Schmidt <thaytan@mad.scientist.com>
*
* gstsunaudiosink.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_SUNAUDIO_SINK_H__
#define __GST_SUNAUDIO_SINK_H__
#include <sys/audioio.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiosink.h>
G_BEGIN_DECLS
#define GST_TYPE_SUNAUDIO_SINK (gst_sunaudiosink_get_type())
#define GST_SUNAUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SUNAUDIO_SINK,GstSunAudioSink))
#define GST_SUNAUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SUNAUDIO_SINK,GstSunAudioSinkClass))
#define GST_IS_SUNAUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SUNAUDIO_SINK))
#define GST_IS_SUNAUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SUNAUDIO_SINK))
typedef struct _GstSunAudioSink GstSunAudioSink;
typedef struct _GstSunAudioSinkClass GstSunAudioSinkClass;
struct _GstSunAudioSink {
GstAudioSink sink;
gchar *device;
gint fd;
audio_device_t dev;
audio_info_t info;
/* Number of segments the ringbuffer is configured for */
guint segtotal;
guint segtotal_samples;
/* Number of segments written to the device */
gint segs_written;
/* Number of samples written to the device */
gint samples_written;
guint bytes_per_sample;
/* mutex and gcond used to control the write method */
GMutex *write_mutex;
GCond *sleep_cond;
gboolean flushing;
};
struct _GstSunAudioSinkClass {
GstAudioSinkClass parent_class;
};
GType gst_sunaudiosink_get_type(void);
G_END_DECLS
#endif /* __GST_SUNAUDIO_SINK_H__ */

421
sys/sunaudio/gstsunaudiosrc.c
View file

@ -1,421 +0,0 @@
/*
* GStreamer - SunAudio source
* Copyright (C) 2005,2006 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
*
* gstsunaudiosrc.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-sunaudiosrc
*
* sunaudiosrc is an audio source designed to work with the Sun Audio
* interface available in Solaris.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 sunaudiosrc ! wavenc ! filesink location=audio.wav
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <fcntl.h>
#include <string.h>
#include <unistd.h>
#include <stropts.h>
#include <sys/mixer.h>
#include "gstsunaudiosrc.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
static void gst_sunaudiosrc_base_init (gpointer g_class);
static void gst_sunaudiosrc_class_init (GstSunAudioSrcClass * klass);
static void gst_sunaudiosrc_init (GstSunAudioSrc * sunaudiosrc,
GstSunAudioSrcClass * g_class);
static void gst_sunaudiosrc_dispose (GObject * object);
static void gst_sunaudiosrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_sunaudiosrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_sunaudiosrc_getcaps (GstBaseSrc * bsrc);
static gboolean gst_sunaudiosrc_open (GstAudioSrc * asrc);
static gboolean gst_sunaudiosrc_close (GstAudioSrc * asrc);
static gboolean gst_sunaudiosrc_prepare (GstAudioSrc * asrc,
GstRingBufferSpec * spec);
static gboolean gst_sunaudiosrc_unprepare (GstAudioSrc * asrc);
static guint gst_sunaudiosrc_read (GstAudioSrc * asrc, gpointer data,
guint length);
static guint gst_sunaudiosrc_delay (GstAudioSrc * asrc);
static void gst_sunaudiosrc_reset (GstAudioSrc * asrc);
#define DEFAULT_DEVICE "/dev/audio"
enum
{
PROP_0,
PROP_DEVICE
};
GST_BOILERPLATE_WITH_INTERFACE (GstSunAudioSrc, gst_sunaudiosrc,
GstAudioSrc, GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_sunaudiosrc);
GST_IMPLEMENT_SUNAUDIO_MIXER_CTRL_METHODS (GstSunAudioSrc, gst_sunaudiosrc);
static GstStaticPadTemplate gst_sunaudiosrc_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16, "
/* [5510,48000] seems to be a Solaris limit */
"rate = (int) [ 5510, 48000 ], " "channels = (int) [ 1, 2 ]")
);
static void
gst_sunaudiosrc_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_sunaudiosrc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_static_pad_template (element_class,
&gst_sunaudiosrc_factory);
gst_element_class_set_static_metadata (element_class, "Sun Audio Source",
"Source/Audio", "Audio source for Sun Audio devices",
"Brian Cameron <brian.cameron@sun.com>");
}
static void
gst_sunaudiosrc_class_init (GstSunAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
gobject_class->dispose = gst_sunaudiosrc_dispose;
gobject_class->get_property = gst_sunaudiosrc_get_property;
gobject_class->set_property = gst_sunaudiosrc_set_property;
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_getcaps);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_open);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_unprepare);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_sunaudiosrc_reset);
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"SunAudio device (usually /dev/audio)", DEFAULT_DEVICE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_sunaudiosrc_init (GstSunAudioSrc * sunaudiosrc,
GstSunAudioSrcClass * g_class)
{
const char *audiodev;
GST_DEBUG_OBJECT (sunaudiosrc, "initializing sunaudiosrc");
sunaudiosrc->fd = -1;
audiodev = g_getenv ("AUDIODEV");
if (audiodev == NULL)
audiodev = DEFAULT_DEVICE;
sunaudiosrc->device = g_strdup (audiodev);
}
static void
gst_sunaudiosrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSunAudioSrc *sunaudiosrc;
sunaudiosrc = GST_SUNAUDIO_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (sunaudiosrc->device);
sunaudiosrc->device = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_sunaudiosrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSunAudioSrc *sunaudiosrc;
sunaudiosrc = GST_SUNAUDIO_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, sunaudiosrc->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_sunaudiosrc_getcaps (GstBaseSrc * bsrc)
{
GstPadTemplate *pad_template;
GstCaps *caps = NULL;
GstSunAudioSrc *sunaudiosrc = GST_SUNAUDIO_SRC (bsrc);
GST_DEBUG_OBJECT (sunaudiosrc, "getcaps called");
pad_template = gst_static_pad_template_get (&gst_sunaudiosrc_factory);
caps = gst_caps_copy (gst_pad_template_get_caps (pad_template));
gst_object_unref (pad_template);
return caps;
}
static gboolean
gst_sunaudiosrc_open (GstAudioSrc * asrc)
{
GstSunAudioSrc *sunaudiosrc = GST_SUNAUDIO_SRC (asrc);
int fd, ret;
fd = open (sunaudiosrc->device, O_RDONLY);
if (fd == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, OPEN_READ, (NULL),
("can't open connection to Sun Audio device %s", sunaudiosrc->device));
return FALSE;
}
sunaudiosrc->fd = fd;
ret = ioctl (fd, AUDIO_GETDEV, &sunaudiosrc->dev);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
GST_DEBUG_OBJECT (sunaudiosrc, "name %s", sunaudiosrc->dev.name);
GST_DEBUG_OBJECT (sunaudiosrc, "version %s", sunaudiosrc->dev.version);
GST_DEBUG_OBJECT (sunaudiosrc, "config %s", sunaudiosrc->dev.config);
ret = ioctl (fd, AUDIO_GETINFO, &sunaudiosrc->info);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
GST_DEBUG_OBJECT (sunaudiosrc, "monitor_gain %d",
sunaudiosrc->info.monitor_gain);
GST_DEBUG_OBJECT (sunaudiosrc, "output_muted %d",
sunaudiosrc->info.output_muted);
GST_DEBUG_OBJECT (sunaudiosrc, "hw_features %08x",
sunaudiosrc->info.hw_features);
GST_DEBUG_OBJECT (sunaudiosrc, "sw_features %08x",
sunaudiosrc->info.sw_features);
GST_DEBUG_OBJECT (sunaudiosrc, "sw_features_enabled %08x",
sunaudiosrc->info.sw_features_enabled);
if (!sunaudiosrc->mixer) {
const char *audiodev;
audiodev = g_getenv ("AUDIODEV");
if (audiodev == NULL) {
sunaudiosrc->mixer = gst_sunaudiomixer_ctrl_new ("/dev/audioctl");
} else {
gchar *device = g_strdup_printf ("%sctl", audiodev);
sunaudiosrc->mixer = gst_sunaudiomixer_ctrl_new (device);
g_free (device);
}
}
return TRUE;
}
static gboolean
gst_sunaudiosrc_close (GstAudioSrc * asrc)
{
GstSunAudioSrc *sunaudiosrc = GST_SUNAUDIO_SRC (asrc);
close (sunaudiosrc->fd);
sunaudiosrc->fd = -1;
if (sunaudiosrc->mixer) {
gst_sunaudiomixer_ctrl_free (sunaudiosrc->mixer);
sunaudiosrc->mixer = NULL;
}
return TRUE;
}
static gboolean
gst_sunaudiosrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
GstSunAudioSrc *sunaudiosrc = GST_SUNAUDIO_SRC (asrc);
audio_info_t ainfo;
int ret;
GstSunAudioMixerCtrl *mixer;
struct audio_info audioinfo;
ret = ioctl (sunaudiosrc->fd, AUDIO_GETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
if (spec->width != 16)
return FALSE;
AUDIO_INITINFO (&ainfo);
ainfo.record.sample_rate = spec->rate;
ainfo.record.precision = spec->width;
ainfo.record.channels = spec->channels;
ainfo.record.encoding = AUDIO_ENCODING_LINEAR;
ainfo.record.buffer_size = spec->buffer_time;
mixer = sunaudiosrc->mixer;
if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
g_warning ("Error getting audio device volume");
}
ainfo.record.port = audioinfo.record.port;
ainfo.record.gain = audioinfo.record.gain;
ainfo.record.balance = audioinfo.record.balance;
spec->segsize = 128;
spec->segtotal = spec->buffer_time / 128;
spec->silence_sample[0] = 0;
spec->silence_sample[1] = 0;
spec->silence_sample[2] = 0;
spec->silence_sample[3] = 0;
ret = ioctl (sunaudiosrc->fd, AUDIO_SETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
ioctl (sunaudiosrc->fd, I_FLUSH, FLUSHR);
return TRUE;
}
static gboolean
gst_sunaudiosrc_unprepare (GstAudioSrc * asrc)
{
return TRUE;
}
static guint
gst_sunaudiosrc_read (GstAudioSrc * asrc, gpointer data, guint length)
{
return read (GST_SUNAUDIO_SRC (asrc)->fd, data, length);
}
static guint
gst_sunaudiosrc_delay (GstAudioSrc * asrc)
{
return 0;
}
static void
gst_sunaudiosrc_reset (GstAudioSrc * asrc)
{
/* Get current values */
GstSunAudioSrc *sunaudiosrc = GST_SUNAUDIO_SRC (asrc);
audio_info_t ainfo;
int ret;
ret = ioctl (sunaudiosrc->fd, AUDIO_GETINFO, &ainfo);
if (ret == -1) {
/*
* Should never happen, but if we couldn't getinfo, then no point
* trying to setinfo
*/
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return;
}
/*
* Pause the audio - so audio stops playing immediately rather than
* waiting for the ringbuffer to empty.
*/
ainfo.record.pause = !NULL;
ret = ioctl (sunaudiosrc->fd, AUDIO_SETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
}
/* Flush the audio */
ret = ioctl (sunaudiosrc->fd, I_FLUSH, FLUSHR);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
}
/* unpause the audio */
ainfo.record.pause = NULL;
ret = ioctl (sunaudiosrc->fd, AUDIO_SETINFO, &ainfo);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosrc, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
}
}

67
sys/sunaudio/gstsunaudiosrc.h
View file

@ -1,67 +0,0 @@
/* GStreamer - SunAudio source
* Copyright (C) 2005,2006 Sun Microsystems, Inc.,
* Brian Cameron <brian.cameron@sun.com>
*
* gstsunaudiosrc.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_SUNAUDIO_SRC_H__
#define __GST_SUNAUDIO_SRC_H__
#include <sys/audioio.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiosrc.h>
#include "gstsunaudiomixerctrl.h"
G_BEGIN_DECLS
#define GST_TYPE_SUNAUDIO_SRC (gst_sunaudiosrc_get_type())
#define GST_SUNAUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SUNAUDIO_SRC,GstSunAudioSrc))
#define GST_SUNAUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SUNAUDIO_SRC,GstSunAudioSrcClass))
#define GST_IS_SUNAUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SUNAUDIO_SRC))
#define GST_IS_SUNAUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SUNAUDIO_SRC))
typedef struct _GstSunAudioSrc GstSunAudioSrc;
typedef struct _GstSunAudioSrcClass GstSunAudioSrcClass;
struct _GstSunAudioSrc {
GstAudioSrc src;
gchar *device;
gint fd;
gint control_fd;
audio_device_t dev;
audio_info_t info;
gint bytes_per_sample;
GstSunAudioMixerCtrl *mixer;
};
struct _GstSunAudioSrcClass {
GstAudioSrcClass parent_class;
};
GType gst_sunaudiosrc_get_type(void);
G_END_DECLS
#endif /* __GST_SUNAUDIO_SRC_H__ */

14
tests/check/Makefile.am
View file

@ -255,12 +255,6 @@ else
check_spectrum =
endif
if USE_SUNAUDIO
check_sunaudio = elements/sunaudio
else
check_sunaudio =
endif
if USE_PLUGIN_SHAPEWIPE
check_shapewipe = elements/shapewipe
else
@ -382,7 +376,6 @@ check_PROGRAMS = \
$(check_shapewipe) \
$(check_soup) \
$(check_spectrum) \
$(check_sunaudio) \
$(check_taglib) \
$(check_udp) \
$(check_videobox) \
@ -552,13 +545,6 @@ elements_rtpmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(G
elements_souphttpsrc_CFLAGS = $(SOUP_CFLAGS) $(AM_CFLAGS)
elements_souphttpsrc_LDADD = $(SOUP_LIBS) $(LDADD)
elements_sunaudio_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) \
$(AM_CFLAGS)
elements_sunaudio_LDADD = \
$(GST_PLUGINS_BASE_LIBS) \
$(LDADD)
elements_udpsrc_CFLAGS = $(AM_CFLAGS) $(GIO_CFLAGS)
elements_udpsrc_LDADD = $(LDADD) $(GIO_LIBS)

1
tests/check/elements/.gitignore vendored
View file

@ -67,7 +67,6 @@ shapewipe
souphttpsrc
spectrum
splitmux
sunaudio
udpsink
udpsrc
videocrop

95
tests/check/elements/sunaudio.c
View file

@ -1,95 +0,0 @@
/* GStreamer unit tests for the sun audio elements
*
* Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <gst/check/gstcheck.h>
#include <gst/interfaces/propertyprobe.h>
#include <gst/interfaces/mixer.h>
#include <gst/gst.h>
GST_START_TEST (test_sun_audio_mixer_track)
{
GstStateChangeReturn state_ret;
GstElement *mixer;
GList *tracks, *l;
mixer = gst_element_factory_make ("sunaudiomixer", "sunaudiomixer");
fail_unless (mixer != NULL, "Failed to create 'sunaudiomixer' element!");
state_ret = gst_element_set_state (mixer, GST_STATE_READY);
if (state_ret != GST_STATE_CHANGE_SUCCESS) {
gst_object_unref (mixer);
return;
}
GST_LOG ("opened sunaudiomixer");
fail_unless (GST_IS_MIXER (mixer), "is not a GstMixer?!");
tracks = (GList *) gst_mixer_list_tracks (GST_MIXER (mixer));
for (l = tracks; l != NULL; l = l->next) {
GObjectClass *klass;
GstMixerTrack *track;
gchar *ulabel = NULL, *label = NULL;
track = GST_MIXER_TRACK (l->data);
g_object_get (track, "label", &label, NULL);
fail_unless (label == NULL || g_utf8_validate (label, -1, NULL));
/* FIXME: remove this check once we depend on -base >= 0.10.12.1 */
klass = G_OBJECT_GET_CLASS (track);
if (g_object_class_find_property (klass, "untranslated-label")) {
g_object_get (track, "untranslated-label", &ulabel, NULL);
}
if (ulabel != NULL) {
gchar *p;
for (p = ulabel; p != NULL && *p != '\0'; ++p) {
fail_unless (g_ascii_isprint (*p),
"untranslated label '%s' not printable ASCII", ulabel);
}
}
GST_DEBUG ("%s: %s", GST_STR_NULL (ulabel), GST_STR_NULL (label));
g_free (label);
g_free (ulabel);
}
fail_unless_equals_int (gst_element_set_state (mixer, GST_STATE_NULL),
GST_STATE_CHANGE_SUCCESS);
gst_object_unref (mixer);
}
GST_END_TEST;
static Suite *
sunaudio_suite (void)
{
Suite *s = suite_create ("sunaudio");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_sun_audio_mixer_track);
return s;
}
GST_CHECK_MAIN (sunaudio)

1
tests/check/meson.build
View file

@ -80,7 +80,6 @@ good_tests = [
[ 'elements/rtpsession' ],
[ 'elements/souphttpsrc', not libsoup_dep.found(), [libsoup_dep] ],
[ 'elements/spectrum' ],
# [ 'elements/sunaudio' ],
[ 'elements/shapewipe' ],
[ 'elements/id3v2mux', not taglib_dep.found() ],
[ 'elements/apev2mux', not taglib_dep.found() ],