gstreamer/sys/sunaudio/gstsunaudiosrc.c
Brian Cameron 23b9485530 sys/sunaudio/: Fix up copyrights (#525860).
Original commit message from CVS:
Patch by: Brian Cameron <brian.cameron at sun dot com>
* sys/sunaudio/gstsunaudio.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiomixer.h:
* sys/sunaudio/gstsunaudiomixerctrl.c:
* sys/sunaudio/gstsunaudiomixerctrl.h:
* sys/sunaudio/gstsunaudiomixertrack.c:
* sys/sunaudio/gstsunaudiomixertrack.h:
* sys/sunaudio/gstsunaudiosink.c:
* sys/sunaudio/gstsunaudiosink.h:
* sys/sunaudio/gstsunaudiosrc.c:
* sys/sunaudio/gstsunaudiosrc.h:
Fix up copyrights (#525860).
2008-04-02 22:37:29 +00:00

433 lines
12 KiB
C

/*
* 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., 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 <sys/mixer.h>
#include "gstsunaudiosrc.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
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"
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_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));
}
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:
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;
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;
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)
{
/* 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)));
}
}