2006-06-20 Wim Taymans <wim@fluendo.com>
+ Patch by: Brian Cameron <brian dot cameron at sun dot com>
+
+ * sys/sunaudio/Makefile.am:
+ * sys/sunaudio/gstsunaudio.c: (plugin_init):
+ * sys/sunaudio/gstsunaudiomixerctrl.c:
+ (gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new),
+ (gst_sunaudiomixer_ctrl_list_tracks),
+ (gst_sunaudiomixer_ctrl_get_volume),
+ (gst_sunaudiomixer_ctrl_set_volume),
+ (gst_sunaudiomixer_ctrl_set_mute),
+ (gst_sunaudiomixer_ctrl_set_record):
+ * sys/sunaudio/gstsunaudiomixerctrl.h:
+ * sys/sunaudio/gstsunaudiomixertrack.c:
+ (gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new):
+ * sys/sunaudio/gstsunaudiomixertrack.h:
+ * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose),
+ (gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init),
+ (gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property),
+ (gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps),
+ (gst_sunaudiosrc_open), (gst_sunaudiosrc_close),
+ (gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare),
+ (gst_sunaudiosrc_read), (gst_sunaudiosrc_delay),
+ (gst_sunaudiosrc_reset):
+ * sys/sunaudio/gstsunaudiosrc.h:
+ Add a SunAudio source plugin.
+ Support stereo and right/left channel gain in the mixer plugin.
+ Support the RECORD flag so that you can switch between line-input and
+ microphone in gnome-volume-control.
+ Code cleanups like using an enumerator for track number instead of an
+ integer. Fixes #344923.
+
+2006-06-20 Wim Taymans <wim@fluendo.com>
+
Patch by: Joni Valtanen <joni dot valtanen at movial dot fi>
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
gstsunaudiosink.c \
gstsunaudiomixerctrl.c \
gstsunaudiomixer.c \
- gstsunaudiomixertrack.c
+ gstsunaudiomixertrack.c \
+ gstsunaudiosrc.c
libgstsunaudio_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
libgstsunaudio_la_LIBADD = \
+ -lgstinterfaces-@GST_MAJORMINOR@ \
+ -lgstaudio-@GST_MAJORMINOR@ \
$(GST_PLUGINS_BASE_LIBS) \
- -lgstinterfaces-$(GST_MAJORMINOR) \
- -lgstaudio-(GST_MAJORMINOR) \
$(GST_LIBS)
libgstsunaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstsunaudiosink.h \
gstsunaudiomixer.h \
gstsunaudiomixerctrl.h \
- gstsunaudiomixertrack.h
+ gstsunaudiomixertrack.h \
+ gstsunaudiosrc.h
#include "gstsunaudiomixer.h"
#include "gstsunaudiosink.h"
+#include "gstsunaudiosrc.h"
extern gchar *__gst_oss_plugin_dir;
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_TYPE_SUNAUDIO_SINK) ||
+ !gst_element_register (plugin, "sunaudiosrc", GST_RANK_SECONDARY,
+ GST_TYPE_SUNAUDIO_SRC)) {
return FALSE;
}
gst_sunaudiomixer_ctrl_build_list (GstSunAudioMixerCtrl * mixer)
{
GstMixerTrack *track;
-
- g_return_if_fail (mixer->mixer_fd != -1);
+ struct audio_info audioinfo;
/*
* Do not continue appending the same 3 static tracks onto the list
*/
if (mixer->tracklist == NULL) {
- track = gst_sunaudiomixer_track_new (0, 1, GST_MIXER_TRACK_OUTPUT);
+ g_return_if_fail (mixer->mixer_fd != -1);
+
+ /* Output */
+ track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_OUTPUT,
+ 2, GST_MIXER_TRACK_OUTPUT);
mixer->tracklist = g_list_append (mixer->tracklist, track);
- track = gst_sunaudiomixer_track_new (1, 1, GST_MIXER_TRACK_INPUT);
+
+ /* Input */
+ track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_LINE_IN,
+ 2, GST_MIXER_TRACK_INPUT);
+
+ /* Set whether we are recording from microphone or from line-in */
+ if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
+ g_warning ("Error getting audio device volume");
+ return;
+ }
+
+ /* Set initial RECORD status */
+ if (audioinfo.record.port == AUDIO_MICROPHONE) {
+ mixer->recdevs |= (1 << GST_SUNAUDIO_TRACK_LINE_IN);
+ track->flags |= GST_MIXER_TRACK_RECORD;
+ } else {
+ mixer->recdevs &= ~(1 << GST_SUNAUDIO_TRACK_LINE_IN);
+ track->flags &= ~GST_MIXER_TRACK_RECORD;
+ }
+
+ /* Monitor */
mixer->tracklist = g_list_append (mixer->tracklist, track);
- track = gst_sunaudiomixer_track_new (2, 1, GST_MIXER_TRACK_OUTPUT);
+ track = gst_sunaudiomixer_track_new (GST_SUNAUDIO_TRACK_MONITOR,
+ 2, GST_MIXER_TRACK_OUTPUT);
mixer->tracklist = g_list_append (mixer->tracklist, track);
}
}
gst_sunaudiomixer_ctrl_list_tracks (GstSunAudioMixerCtrl * mixer)
{
gst_sunaudiomixer_ctrl_build_list (mixer);
+
return (const GList *) mixer->tracklist;
}
gst_sunaudiomixer_ctrl_get_volume (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track, gint * volumes)
{
+ gint gain, balance;
+ float ratio;
struct audio_info audioinfo;
GstSunAudioMixerTrack *sunaudiotrack = GST_SUNAUDIO_MIXER_TRACK (track);
}
switch (sunaudiotrack->track_num) {
- case 0:
- sunaudiotrack->vol = volumes[0] =
- (audioinfo.play.gain / SCALE_FACTOR) + 0.5;
+ case GST_SUNAUDIO_TRACK_OUTPUT:
+ gain = (int) ((float) audioinfo.play.gain / (float) SCALE_FACTOR + 0.5);
+ balance = audioinfo.play.balance;
break;
- case 1:
- sunaudiotrack->vol = volumes[0] =
- (audioinfo.record.gain / SCALE_FACTOR) + 0.5;
+ case GST_SUNAUDIO_TRACK_LINE_IN:
+ gain = (int) ((float) audioinfo.record.gain / (float) SCALE_FACTOR + 0.5);
+ balance = audioinfo.record.balance;
break;
- case 2:
- sunaudiotrack->vol = volumes[0] =
- (audioinfo.monitor_gain / SCALE_FACTOR) + 0.5;
+ case GST_SUNAUDIO_TRACK_MONITOR:
+ gain =
+ (int) ((float) audioinfo.monitor_gain / (float) SCALE_FACTOR + 0.5);
+ balance = audioinfo.record.balance;
break;
}
+
+ 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);
+ }
+
+ /*
+ * Reset whether we are recording from microphone or from line-in.
+ * This can change if another program resets the value (such as
+ * sdtaudiocontrol), so it is good to update the flag when we
+ * get the volume. The gnome-volume-control program calls this
+ * function in a loop so the value will update properly when
+ * changed.
+ */
+ if ((audioinfo.record.port == AUDIO_MICROPHONE &&
+ !GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) ||
+ (audioinfo.record.port == AUDIO_LINE_IN &&
+ GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD))) {
+
+ if (audioinfo.record.port == AUDIO_MICROPHONE) {
+ mixer->recdevs |= (1 << GST_SUNAUDIO_TRACK_LINE_IN);
+ track->flags |= GST_MIXER_TRACK_RECORD;
+ } else {
+ mixer->recdevs &= ~(1 << GST_SUNAUDIO_TRACK_LINE_IN);
+ track->flags &= ~GST_MIXER_TRACK_RECORD;
+ }
+ }
+
+ /* Likewise reset MUTE */
+ if ((sunaudiotrack->track_num == GST_SUNAUDIO_TRACK_OUTPUT &&
+ audioinfo.output_muted == 1) ||
+ (sunaudiotrack->track_num != GST_SUNAUDIO_TRACK_OUTPUT && gain == 0)) {
+ track->flags |= GST_MIXER_TRACK_MUTE;
+ } else {
+ /*
+ * If MUTE is set, then gain is always 0, so don't bother
+ * resetting our internal value.
+ */
+ 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 volume;
+ gint gain;
+ gint balance;
+ gint l_real_gain;
+ gint r_real_gain;
+ float ratio;
+ gchar buf[100];
struct audio_info audioinfo;
GstSunAudioMixerTrack *sunaudiotrack = GST_SUNAUDIO_MIXER_TRACK (track);
+ gint temp[2];
+
+ l_real_gain = volumes[0];
+ r_real_gain = volumes[1];
+
+ if (l_real_gain == r_real_gain) {
+ gain = (int) ((float) l_real_gain * (float) SCALE_FACTOR + 0.5);
+ balance = AUDIO_MID_BALANCE;
+ } else if (l_real_gain < r_real_gain) {
+ gain = (int) ((float) r_real_gain * (float) SCALE_FACTOR + 0.5);
+ ratio = (float) l_real_gain / (float) r_real_gain;
+ balance =
+ AUDIO_RIGHT_BALANCE - (int) (ratio * (float) AUDIO_MID_BALANCE + 0.5);
+ } else {
+ gain = (int) ((float) l_real_gain * (float) SCALE_FACTOR + 0.5);
+ ratio = (float) r_real_gain / (float) l_real_gain;
+ balance =
+ AUDIO_LEFT_BALANCE + (int) (ratio * (float) AUDIO_MID_BALANCE + 0.5);
+ }
- g_return_if_fail (mixer->mixer_fd != -1);
-
- if (volume < 0)
- volume = 0;
+ sunaudiotrack->gain = gain;
+ sunaudiotrack->balance = balance;
- volume = volumes[0] * SCALE_FACTOR;
+ if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_MUTE))
+ return;
/* Set the volume */
AUDIO_INITINFO (&audioinfo);
switch (sunaudiotrack->track_num) {
- case 0:
- audioinfo.play.gain = volume;
+ case GST_SUNAUDIO_TRACK_OUTPUT:
+ audioinfo.play.gain = gain;
+ audioinfo.play.balance = balance;
break;
- case 1:
- audioinfo.record.gain = volume;
+ case GST_SUNAUDIO_TRACK_LINE_IN:
+ audioinfo.record.gain = gain;
+ audioinfo.record.balance = balance;
break;
- case 2:
- audioinfo.monitor_gain = volume;
+ case GST_SUNAUDIO_TRACK_MONITOR:
+ audioinfo.monitor_gain = gain;
+ audioinfo.record.balance = balance;
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;
}
-
- sunaudiotrack->vol = volume;
}
void
{
struct audio_info audioinfo;
GstSunAudioMixerTrack *sunaudiotrack = GST_SUNAUDIO_MIXER_TRACK (track);
- gint volume;
-
- g_return_if_fail (mixer->mixer_fd != -1);
+ gint volume, balance;
AUDIO_INITINFO (&audioinfo);
volume = 0;
track->flags |= GST_MIXER_TRACK_MUTE;
} else {
- volume = sunaudiotrack->vol;
+ volume = sunaudiotrack->gain;
track->flags &= ~GST_MIXER_TRACK_MUTE;
}
+ balance = sunaudiotrack->balance;
+
switch (sunaudiotrack->track_num) {
- case 0:
+ case GST_SUNAUDIO_TRACK_OUTPUT:
if (mute)
audioinfo.output_muted = 1;
audioinfo.output_muted = 0;
audioinfo.play.gain = volume;
+ audioinfo.play.balance = balance;
break;
- case 1:
+ case GST_SUNAUDIO_TRACK_LINE_IN:
audioinfo.record.gain = volume;
+ audioinfo.record.balance = balance;
break;
- case 2:
+ case GST_SUNAUDIO_TRACK_MONITOR:
audioinfo.monitor_gain = volume;
+ audioinfo.record.balance = balance;
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;
gst_sunaudiomixer_ctrl_set_record (GstSunAudioMixerCtrl * mixer,
GstMixerTrack * track, gboolean record)
{
- /* Implementation Pending */
+ GstSunAudioMixerTrack *sunaudiotrack = GST_SUNAUDIO_MIXER_TRACK (track);
+ struct audio_info audioinfo;
+ GList *trk;
+
+ /* Don't change the setting */
+ if ((record && GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) ||
+ (!record && !GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)))
+ return;
+
+ /*
+ * So there is probably no need to look for others, but reset them all
+ * to being off.
+ */
+ for (trk = mixer->tracklist; trk != NULL; trk = trk->next) {
+ GstMixerTrack *turn = (GstMixerTrack *) trk->data;
+
+ turn->flags &= ~GST_MIXER_TRACK_RECORD;
+ }
+ mixer->recdevs = 0;
+
+ /* Set the port */
+ AUDIO_INITINFO (&audioinfo);
+
+ if (record) {
+ audioinfo.record.port = AUDIO_MICROPHONE;
+ mixer->recdevs |= (1 << sunaudiotrack->track_num);
+ track->flags |= GST_MIXER_TRACK_RECORD;
+ } else {
+ audioinfo.record.port = AUDIO_LINE_IN;
+ mixer->recdevs &= ~(1 << sunaudiotrack->track_num);
+ track->flags &= ~GST_MIXER_TRACK_RECORD;
+ }
+
+ 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;
+ }
}
gint mixer_fd;
gchar * device;
+ gint recdevs;
};
GstSunAudioMixerCtrl* gst_sunaudiomixer_ctrl_new (const gchar *device);
#include <string.h>
#include <errno.h>
#include <sys/ioctl.h>
+#include <sys/audioio.h>
#include <gst/gst-i18n-plugin.h>
#include "gstsunaudiomixertrack.h"
-#define MIXER_DEVICES 3
#define MASK_BIT_IS_SET(mask, bit) \
(mask & (1 << bit))
G_DEFINE_TYPE (GstSunAudioMixerTrack, gst_sunaudiomixer_track,
- GST_TYPE_MIXER_TRACK);
+ GST_TYPE_MIXER_TRACK)
-static void
-gst_sunaudiomixer_track_class_init (GstSunAudioMixerTrackClass * klass)
+ static void
+ gst_sunaudiomixer_track_class_init (GstSunAudioMixerTrackClass * klass)
{
/* nop */
}
static void
gst_sunaudiomixer_track_init (GstSunAudioMixerTrack * track)
{
- track->vol = 0;
+ track->gain = 0;
+ track->balance = AUDIO_MID_BALANCE;
track->track_num = 0;
}
}
GstMixerTrack *
-gst_sunaudiomixer_track_new (gint track_num, gint max_chans, gint flags)
+gst_sunaudiomixer_track_new (GstSunAudioTrackType track_num,
+ gint max_chans, gint flags)
{
GstSunAudioMixerTrack *sunaudiotrack;
GstMixerTrack *track;
+ gint volume;
if (!labels)
fill_labels ();
track->min_volume = 0;
track->max_volume = 100;
sunaudiotrack->track_num = track_num;
-
- sunaudiotrack->vol = (0 & 0xff);
+ sunaudiotrack->gain = (0 & 0xff);
+ sunaudiotrack->balance = AUDIO_MID_BALANCE;
return track;
}
G_BEGIN_DECLS
+typedef enum
+{
+ GST_SUNAUDIO_TRACK_OUTPUT = 0,
+ GST_SUNAUDIO_TRACK_LINE_IN = 1,
+ GST_SUNAUDIO_TRACK_MONITOR = 2,
+} GstSunAudioTrackType;
+
+#define MIXER_DEVICES 3
+
#define GST_TYPE_SUNAUDIO_MIXER_TRACK \
(gst_sunaudiomixer_track_get_type ())
#define GST_SUNAUDIO_MIXER_TRACK(obj) \
typedef struct _GstSunAudioMixerTrack {
GstMixerTrack parent;
- gint vol;
- gint track_num;
+ gint gain;
+ gint balance;
+ GstSunAudioTrackType track_num;
} GstSunAudioMixerTrack;
typedef struct _GstSunAudioMixerTrackClass {
--- /dev/null
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ * 2000,2005 Wim Taymans <wim@fluendo.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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-sunaudiosrc
+ *
+ * <refsect2>
+ * <para>
+ * sunaudiosrc is an audio source designed to work with the Sun Audio
+ * interface available in Solaris.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ *
+ * gst-launch sunaudiosrc ! filesink location=outfile
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <fcntl.h>
+#include <string.h>
+#include <unistd.h>
+#include <stropts.h>
+
+#include "gstsunaudiosrc.h"
+
+static GstElementDetails plugin_details =
+GST_ELEMENT_DETAILS ("Sun Audio Source",
+ "Source/Audio",
+ "Audio source for Sun Audio devices",
+ "Brian Cameron <brian.cameron@sun.com>");
+
+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"
+#define DEFAULT_DEVICE_NAME ""
+
+enum
+{
+ PROP_0,
+ PROP_DEVICE,
+ PROP_DEVICE_NAME,
+};
+
+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_pad_template (element_class,
+ gst_static_pad_template_get (&gst_sunaudiosrc_factory));
+ gst_element_class_set_details (element_class, &plugin_details);
+}
+
+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_DEBUG_FUNCPTR (gst_sunaudiosrc_dispose);
+ gobject_class->get_property =
+ GST_DEBUG_FUNCPTR (gst_sunaudiosrc_get_property);
+ gobject_class->set_property =
+ GST_DEBUG_FUNCPTR (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_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+ g_param_spec_string ("device-name", "Device name",
+ "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+ G_PARAM_READABLE));
+}
+
+static void
+gst_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);
+
+ sunaudiosrc->device_name = g_strdup (DEFAULT_DEVICE_NAME);
+}
+
+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:
+ if (sunaudiosrc->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;
+ case PROP_DEVICE_NAME:
+ g_value_set_string (value, sunaudiosrc->device_name);
+ 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_RDWR);
+
+ 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 ("/dev/%sctl", audiodev);
+
+ sunaudiosrc->mixer = gst_sunaudiomixer_ctrl_new (device);
+ g_free (device);
+ }
+ sunaudiosrc->device_name = g_strdup (sunaudiosrc->mixer->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;
+ int ctrl_fd = -1;
+ int ports;
+
+ 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;
+
+ GstSunAudioMixerCtrl *mixer = sunaudiosrc->mixer;
+ struct audio_info audioinfo;
+
+ 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)
+{
+ return;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2004 David A. Schleef <ds@schleef.org>
+ * Copyright (C) 2005 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, 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(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SUNAUDIO_SRC))
+
+typedef struct _GstSunAudioSrc GstSunAudioSrc;
+typedef struct _GstSunAudioSrcClass GstSunAudioSrcClass;
+
+struct _GstSunAudioSrc {
+ GstAudioSrc src;
+
+ gchar *device;
+ gchar *device_name;
+ 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__ */
+
projects / platform / upstream / gst-plugins-good.git / commitdiff