gstreamer/sys/sunaudio/gstsunaudiosink.c

394 lines
11 KiB
C
Raw Normal View History

/*
* GStreamer
* Copyright (C) 2004 David A. Schleef <ds@schleef.org>
* Copyright (C) 2005 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-sunaudiosink
*
* <refsect2>
* <para>
* sunaudiosink is an audio sink designed to work with the Sun Audio
* interface available in Solaris.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch -v sinesrc ! sunaudiosink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <fcntl.h>
#include <string.h>
#include <unistd.h>
#include <sys/mman.h>
#include "gstsunaudiosink.h"
/* elementfactory information */
Define GstElementDetails as const and also static (when defined as global) Original commit message from CVS: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init): * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: (gst_shout2send_init): * ext/shout2/gstshout2.h: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): * gst/avi/gstavidemux.c: (gst_avi_demux_base_init): * gst/avi/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.c: * gst/debug/gstnavseek.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/law/alaw-decode.c: (gst_alawdec_base_init): * gst/law/alaw-encode.c: (gst_alawenc_base_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init): * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init): * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init): * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstaggregator.c: * gst/oldcore/gstfdsink.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstshaper.c: * gst/oldcore/gststatistics.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: (gst_gamma_base_init): * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: (gst_videotemplate_base_init): * gst/videomixer/videomixer.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_dispose), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_send_event), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_chain), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state): * gst/wavparse/gstwavparse.h: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: Define GstElementDetails as const and also static (when defined as global)
2006-04-25 21:39:46 +00:00
static const GstElementDetails plugin_details =
GST_ELEMENT_DETAILS ("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_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_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_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_sunaudiosink_factory));
gst_element_class_set_details (element_class, &plugin_details);
}
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;
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_class_init): * ext/esd/esdsink.c: (gst_esdsink_class_init): * ext/flac/gstflactag.c: (gst_flac_tag_class_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init): * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init): * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init): * ext/libmng/gstmngdec.c: (gst_mngdec_class_init): * ext/libmng/gstmngenc.c: (gst_mngenc_class_init): * ext/libpng/gstpngdec.c: (gst_pngdec_class_init): * ext/libpng/gstpngenc.c: (gst_pngenc_class_init): * ext/mikmod/gstmikmod.c: (gst_mikmod_class_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init): * ext/speex/gstspeexenc.c: (gst_speexenc_class_init): * gst/alpha/gstalpha.c: (gst_alpha_class_init): * gst/avi/gstavimux.c: (gst_avimux_class_init): * gst/debug/efence.c: (gst_efence_class_init): * gst/debug/negotiation.c: (gst_negotiation_class_init): * gst/flx/gstflxdec.c: (gst_flxdec_class_init): * gst/goom/gstgoom.c: (gst_goom_class_init): * gst/id3demux/gstid3demux.c: (gst_id3demux_class_init): * gst/interleave/deinterleave.c: (deinterleave_class_init): * gst/interleave/interleave.c: (interleave_class_init): * gst/law/alaw-decode.c: (gst_alawdec_class_init): * gst/law/alaw-encode.c: (gst_alawenc_class_init): * gst/law/mulaw-encode.c: (gst_mulawenc_class_init): * gst/median/gstmedian.c: (gst_median_class_init): * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init): * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init): * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init): * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init): * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init): * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init): * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init): * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init): * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init): * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init): * gst/smpte/gstsmpte.c: (gst_smpte_class_init): * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init): * gst/udp/gstudpsink.c: (gst_udpsink_class_init): * gst/videomixer/videomixer.c: (gst_videomixer_class_init): * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init): * sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init): * sys/oss/gstosssink.c: (gst_oss_sink_class_init): * sys/osxaudio/gstosxaudioelement.c: (gst_osxaudioelement_class_init): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init): * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:21:45 +00:00
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_sunaudiosink_dispose;
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_sunaudiosink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (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));
}
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);
}
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:
g_free (sunaudiosink->device);
sunaudiosink->device = g_strdup (g_value_get_string (value));
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:
g_value_set_string (value, sunaudiosink->device);
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 */
fd = open (sunaudiosink->device, O_WRONLY | O_NONBLOCK);
if (fd >= 0) {
close (fd);
fd = open (sunaudiosink->device, O_WRONLY);
}
if (fd == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, OPEN_WRITE, (NULL),
("can't open connection to Sun Audio device %s", sunaudiosink->device));
return FALSE;
}
sunaudiosink->fd = fd;
ret = ioctl (fd, AUDIO_GETDEV, &sunaudiosink->dev);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
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) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
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;
}
static gboolean
gst_sunaudiosink_close (GstAudioSink * asink)
{
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
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;
/*
* SunAudio doesn't really give access to buffer size, these values work. Setting
* the buffer so large (512K) is a bit annoying because this causes the volume
* control in audio players to be slow in responding since the audio volume won't
* change until the buffer empties. SunAudio doesn't seem to allow changing the
* audio output buffer size to anything smaller, though. I notice setting the
* values smaller causes the audio to stutter, which is worse.
*/
spec->segsize = 4096;
spec->segtotal = 128;
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;
}
return TRUE;
}
static gboolean
gst_sunaudiosink_unprepare (GstAudioSink * asink)
{
return TRUE;
}
static guint
gst_sunaudiosink_write (GstAudioSink * asink, gpointer data, guint length)
{
return write (GST_SUNAUDIO_SINK (asink)->fd, data, length);
}
/*
* Should provide the current delay between writing a sample to the
* audio device and that sample being actually played. Returning 0 for
* now, but this isn't good for synchronization
*/
static guint
gst_sunaudiosink_delay (GstAudioSink * asink)
{
return 0;
}
static void
gst_sunaudiosink_reset (GstAudioSink * asink)
{
}