gstreamer/sys/oss/gstosssink.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* gstosssink.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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#ifdef HAVE_OSS_INCLUDE_IN_SYS
# include <sys/soundcard.h>
#else
# ifdef HAVE_OSS_INCLUDE_IN_ROOT
# include <soundcard.h>
# else
# ifdef HAVE_OSS_INCLUDE_IN_MACHINE
# include <machine/soundcard.h>
# else
# error "What to include?"
# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include "common.h"
#include "gstosssink.h"
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
/* 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 gst_oss_sink_details =
GST_ELEMENT_DETAILS ("Audio Sink (OSS)",
"Sink/Audio",
"Output to a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Wim Taymans <wim.taymans@chello.be>");
static void gst_oss_sink_base_init (gpointer g_class);
static void gst_oss_sink_class_init (GstOssSinkClass * klass);
static void gst_oss_sink_init (GstOssSink * osssink);
static void gst_oss_sink_dispose (GObject * object);
static void gst_oss_sink_finalise (GObject * object);
static void gst_oss_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_oss_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink);
static gboolean gst_oss_sink_open (GstAudioSink * asink);
static gboolean gst_oss_sink_close (GstAudioSink * asink);
static gboolean gst_oss_sink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);
static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_oss_sink_delay (GstAudioSink * asink);
static void gst_oss_sink_reset (GstAudioSink * asink);
/* OssSink signals and args */
enum
{
LAST_SIGNAL
};
#define DEFAULT_DEVICE "/dev/dsp"
enum
{
PROP_0,
PROP_DEVICE,
};
static GstStaticPadTemplate osssink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
static GstElementClass *parent_class = NULL;
/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_oss_sink_get_type (void)
{
static GType osssink_type = 0;
if (!osssink_type) {
static const GTypeInfo osssink_info = {
sizeof (GstOssSinkClass),
gst_oss_sink_base_init,
NULL,
(GClassInitFunc) gst_oss_sink_class_init,
NULL,
NULL,
sizeof (GstOssSink),
0,
(GInstanceInitFunc) gst_oss_sink_init,
};
osssink_type =
g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink",
&osssink_info, 0);
}
return osssink_type;
}
static void
gst_oss_sink_dispose (GObject * object)
{
GstOssSink *osssink = GST_OSSSINK (object);
if (osssink->probed_caps) {
gst_caps_unref (osssink->probed_caps);
osssink->probed_caps = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_oss_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_oss_sink_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&osssink_sink_factory));
}
static void
gst_oss_sink_class_init (GstOssSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
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_DEBUG_FUNCPTR (gst_oss_sink_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_oss_sink_finalise);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss_sink_get_property);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss_sink_set_property);
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"OSS device (usually /dev/dspN)", DEFAULT_DEVICE, G_PARAM_READWRITE));
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);
}
static void
gst_oss_sink_init (GstOssSink * osssink)
{
GST_DEBUG_OBJECT (osssink, "initializing osssink");
osssink->device = g_strdup (DEFAULT_DEVICE);
osssink->fd = -1;
}
static void
gst_oss_sink_finalise (GObject * object)
{
GstOssSink *osssink = GST_OSSSINK (object);
g_free (osssink->device);
}
static void
gst_oss_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOssSink *sink;
sink = GST_OSSSINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (sink->device);
sink->device = g_value_dup_string (value);
if (sink->probed_caps) {
gst_caps_unref (sink->probed_caps);
sink->probed_caps = NULL;
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_oss_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOssSink *sink;
sink = GST_OSSSINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, sink->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_oss_sink_getcaps (GstBaseSink * bsink)
{
GstOssSink *osssink;
GstCaps *caps;
osssink = GST_OSSSINK (bsink);
if (osssink->fd == -1) {
caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
(bsink)));
} else if (osssink->probed_caps) {
caps = gst_caps_copy (osssink->probed_caps);
} else {
caps = gst_oss_helper_probe_caps (osssink->fd);
if (caps && !gst_caps_is_empty (caps)) {
osssink->probed_caps = gst_caps_copy (caps);
}
}
return caps;
}
static gint
ilog2 (gint x)
{
/* well... hacker's delight explains... */
x = x | (x >> 1);
x = x | (x >> 2);
x = x | (x >> 4);
x = x | (x >> 8);
x = x | (x >> 16);
x = x - ((x >> 1) & 0x55555555);
x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
x = (x + (x >> 4)) & 0x0f0f0f0f;
x = x + (x >> 8);
x = x + (x >> 16);
return (x & 0x0000003f) - 1;
}
static gint
gst_oss_sink_get_format (GstBufferFormat fmt)
{
gint result;
switch (fmt) {
case GST_MU_LAW:
result = AFMT_MU_LAW;
break;
case GST_A_LAW:
result = AFMT_A_LAW;
break;
case GST_IMA_ADPCM:
result = AFMT_IMA_ADPCM;
break;
case GST_U8:
result = AFMT_U8;
break;
case GST_S16_LE:
result = AFMT_S16_LE;
break;
case GST_S16_BE:
result = AFMT_S16_BE;
break;
case GST_S8:
result = AFMT_S8;
break;
case GST_U16_LE:
result = AFMT_U16_LE;
break;
case GST_U16_BE:
result = AFMT_U16_BE;
break;
case GST_MPEG:
result = AFMT_MPEG;
break;
default:
result = 0;
break;
}
return result;
}
static gboolean
gst_oss_sink_open (GstAudioSink * asink)
{
GstOssSink *oss;
int mode;
oss = GST_OSSSINK (asink);
mode = O_WRONLY;
mode |= O_NONBLOCK;
oss->fd = open (oss->device, mode, 0);
if (oss->fd == -1) {
switch (errno) {
case EBUSY:
goto busy;
default:
goto open_failed;
}
}
return TRUE;
busy:
{
GST_ELEMENT_ERROR (oss, RESOURCE, BUSY, (NULL), (NULL));
return FALSE;
}
open_failed:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE, (NULL), GST_ERROR_SYSTEM);
return FALSE;
}
}
static gboolean
gst_oss_sink_close (GstAudioSink * asink)
{
close (GST_OSSSINK (asink)->fd);
GST_OSSSINK (asink)->fd = -1;
return TRUE;
}
static gboolean
gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstOssSink *oss;
struct audio_buf_info info;
int mode;
int tmp;
oss = GST_OSSSINK (asink);
mode = fcntl (oss->fd, F_GETFL);
mode &= ~O_NONBLOCK;
if (fcntl (oss->fd, F_SETFL, mode) == -1)
goto non_block;
tmp = gst_oss_sink_get_format (spec->format);
if (tmp == 0)
goto wrong_format;
if (spec->width != 16 && spec->width != 8)
goto dodgy_width;
SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT");
if (spec->channels == 2)
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
tmp = ilog2 (spec->segsize);
tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
spec->segsize, spec->segtotal, tmp);
SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE");
spec->segsize = info.fragsize;
spec->segtotal = info.fragstotal;
spec->bytes_per_sample = (spec->width / 8) * spec->channels;
oss->bytes_per_sample = (spec->width / 8) * spec->channels;
memset (spec->silence_sample, 0, spec->bytes_per_sample);
GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
spec->segsize, spec->segtotal, tmp);
return TRUE;
non_block:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("Unable to set device %s in non blocking mode: %s",
oss->device, g_strerror (errno)));
return FALSE;
}
wrong_format:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("Unable to get format %d", spec->format));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("unexpected width %d", spec->width));
return FALSE;
}
}
static gboolean
gst_oss_sink_unprepare (GstAudioSink * asink)
{
/* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
if (!gst_oss_sink_close (asink))
goto couldnt_close;
if (!gst_oss_sink_open (asink))
goto couldnt_reopen;
return TRUE;
couldnt_close:
{
GST_DEBUG_OBJECT (asink, "Could not close the audio device");
return FALSE;
}
couldnt_reopen:
{
GST_DEBUG_OBJECT (asink, "Could not reopen the audio device");
return FALSE;
}
}
static guint
gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
return write (GST_OSSSINK (asink)->fd, data, length);
}
static guint
gst_oss_sink_delay (GstAudioSink * asink)
{
GstOssSink *oss;
gint delay = 0;
gint ret;
oss = GST_OSSSINK (asink);
#ifdef SNDCTL_DSP_GETODELAY
ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
#else
ret = -1;
#endif
if (ret < 0) {
audio_buf_info info;
ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
}
return delay / oss->bytes_per_sample;
}
static void
gst_oss_sink_reset (GstAudioSink * asink)
{
#if 0
GstOssSink *oss;
gint ret;
oss = GST_OSSSINK (asink);
/* deadlocks on my machine... */
ret = ioctl (oss->fd, SNDCTL_DSP_RESET, 0);
#endif
}