mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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91a716f915
Original commit message from CVS: - Changed plugins for new APIs - modularized audiofile. - added seeking, query and convert functions for mad, mpeg2dec, avidemux, mpegdemux, mpegparse - sync updates to oss. removed the ossclock for now
719 lines
21 KiB
C
719 lines
21 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wim.taymans@chello.be>
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*
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* gstosssink.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <sys/ioctl.h>
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#include <fcntl.h>
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#include <sys/soundcard.h>
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#include <unistd.h>
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#include <errno.h>
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#include <gstosssink.h>
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static GstElementDetails gst_osssink_details = {
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"Audio Sink (OSS)",
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"Sink/Audio",
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"Output to a sound card via OSS",
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VERSION,
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"Erik Walthinsen <omega@cse.ogi.edu>, "
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"Wim Taymans <wim.taymans@chello.be>",
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"(C) 1999",
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};
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static void gst_osssink_class_init (GstOssSinkClass *klass);
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static void gst_osssink_init (GstOssSink *osssink);
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static void gst_osssink_finalize (GObject *object);
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static gboolean gst_osssink_open_audio (GstOssSink *sink);
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static void gst_osssink_close_audio (GstOssSink *sink);
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static gboolean gst_osssink_sync_parms (GstOssSink *osssink);
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static GstElementStateReturn gst_osssink_change_state (GstElement *element);
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static void gst_osssink_set_clock (GstElement *element, GstClock *clock);
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//static GstClock* gst_osssink_get_clock (GstElement *element);
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static GstPadConnectReturn gst_osssink_sinkconnect (GstPad *pad, GstCaps *caps);
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static void gst_osssink_set_property (GObject *object, guint prop_id, const GValue *value,
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GParamSpec *pspec);
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static void gst_osssink_get_property (GObject *object, guint prop_id, GValue *value,
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GParamSpec *pspec);
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static void gst_osssink_chain (GstPad *pad,GstBuffer *buf);
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/* OssSink signals and args */
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enum {
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SIGNAL_HANDOFF,
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LAST_SIGNAL
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};
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enum {
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ARG_0,
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ARG_DEVICE,
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ARG_MUTE,
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ARG_FORMAT,
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ARG_CHANNELS,
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ARG_FREQUENCY,
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ARG_FRAGMENT,
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ARG_BUFFER_SIZE
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/* FILL ME */
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};
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GST_PAD_TEMPLATE_FACTORY (osssink_sink_factory,
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"osssink_sink",
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"audio/raw",
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"format", GST_PROPS_STRING ("int"), /* hack */
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"law", GST_PROPS_INT (0),
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_LIST (
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GST_PROPS_BOOLEAN (FALSE),
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GST_PROPS_BOOLEAN (TRUE)
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),
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"width", GST_PROPS_LIST (
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GST_PROPS_INT (8),
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GST_PROPS_INT (16)
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),
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"depth", GST_PROPS_LIST (
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GST_PROPS_INT (8),
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GST_PROPS_INT (16)
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),
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"rate", GST_PROPS_INT_RANGE (1000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2)
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)
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);
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#define GST_TYPE_OSSSINK_CHANNELS (gst_osssink_channels_get_type())
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static GType
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gst_osssink_channels_get_type(void) {
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static GType osssink_channels_type = 0;
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static GEnumValue osssink_channels[] = {
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{0, "0", "Silence"},
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{1, "1", "Mono"},
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{2, "2", "Stereo"},
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{0, NULL, NULL},
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};
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if (!osssink_channels_type) {
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osssink_channels_type = g_enum_register_static("GstAudiosinkChannels", osssink_channels);
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}
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return osssink_channels_type;
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}
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static GstElementClass *parent_class = NULL;
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static guint gst_osssink_signals[LAST_SIGNAL] = { 0 };
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GType
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gst_osssink_get_type (void)
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{
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static GType osssink_type = 0;
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if (!osssink_type) {
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static const GTypeInfo osssink_info = {
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sizeof(GstOssSinkClass),
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NULL,
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NULL,
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(GClassInitFunc)gst_osssink_class_init,
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NULL,
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NULL,
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sizeof(GstOssSink),
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0,
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(GInstanceInitFunc)gst_osssink_init,
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};
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osssink_type = g_type_register_static (GST_TYPE_ELEMENT, "GstOssSink", &osssink_info, 0);
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}
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return osssink_type;
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}
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static GstBufferPool*
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gst_osssink_get_bufferpool (GstPad *pad)
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{
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GstOssSink *oss;
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oss = GST_OSSSINK (gst_pad_get_parent(pad));
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return oss->sinkpool;
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}
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static void
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gst_osssink_finalize (GObject *object)
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{
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GstOssSink *osssink = (GstOssSink *) object;
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g_free (osssink->device);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_osssink_class_init (GstOssSinkClass *klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass*)klass;
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gstelement_class = (GstElementClass*)klass;
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parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_DEVICE,
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g_param_spec_string("device","device","device",
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"/dev/dsp",G_PARAM_READWRITE)); /* CHECKME! */
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MUTE,
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g_param_spec_boolean("mute","mute","mute",
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TRUE,G_PARAM_READWRITE));
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/* it would be nice to show format in symbolic form, oh well */
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FORMAT,
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g_param_spec_int ("format","format","format",
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0, G_MAXINT, AFMT_S16_LE, G_PARAM_READWRITE));
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_CHANNELS,
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g_param_spec_enum("channels","channels","channels",
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GST_TYPE_OSSSINK_CHANNELS,2,G_PARAM_READWRITE));
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY,
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g_param_spec_int("frequency","frequency","frequency",
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0,G_MAXINT,44100,G_PARAM_READWRITE));
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FRAGMENT,
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g_param_spec_int("fragment","fragment","fragment",
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0,G_MAXINT,6,G_PARAM_READWRITE));
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BUFFER_SIZE,
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g_param_spec_int("buffer_size","buffer_size","buffer_size",
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0,G_MAXINT,4096,G_PARAM_READWRITE));
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gst_osssink_signals[SIGNAL_HANDOFF] =
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g_signal_new("handoff",G_TYPE_FROM_CLASS(klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET(GstOssSinkClass,handoff), NULL, NULL,
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g_cclosure_marshal_VOID__VOID,G_TYPE_NONE,0);
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gobject_class->set_property = gst_osssink_set_property;
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gobject_class->get_property = gst_osssink_get_property;
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gobject_class->finalize = gst_osssink_finalize;
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_osssink_change_state);
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}
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static void
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gst_osssink_init (GstOssSink *osssink)
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{
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osssink->sinkpad = gst_pad_new_from_template (
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GST_PAD_TEMPLATE_GET (osssink_sink_factory), "sink");
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gst_element_add_pad (GST_ELEMENT (osssink), osssink->sinkpad);
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gst_pad_set_connect_function (osssink->sinkpad, gst_osssink_sinkconnect);
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gst_pad_set_bufferpool_function (osssink->sinkpad, gst_osssink_get_bufferpool);
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gst_pad_set_chain_function (osssink->sinkpad, gst_osssink_chain);
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osssink->device = g_strdup ("/dev/dsp");
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osssink->fd = -1;
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osssink->channels = 1;
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osssink->frequency = 11025;
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osssink->fragment = 6;
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/* AFMT_*_BE not available on all OSS includes (e.g. FBSD) */
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#ifdef WORDS_BIGENDIAN
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osssink->format = AFMT_S16_BE;
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#else
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osssink->format = AFMT_S16_LE;
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#endif /* WORDS_BIGENDIAN */
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osssink->bufsize = 4096;
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osssink->bps = 0;
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osssink->resync = FALSE;
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/* 6 buffers per chunk by default */
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osssink->sinkpool = gst_buffer_pool_get_default (osssink->bufsize, 6);
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GST_ELEMENT (osssink)->setclockfunc = gst_osssink_set_clock;
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GST_FLAG_SET (osssink, GST_ELEMENT_THREAD_SUGGESTED);
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GST_FLAG_SET (osssink, GST_ELEMENT_EVENT_AWARE);
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}
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static GstPadConnectReturn
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gst_osssink_sinkconnect (GstPad *pad, GstCaps *caps)
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{
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gint law, endianness, width, depth;
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gboolean sign;
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gint format = -1;
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GstOssSink *osssink = GST_OSSSINK (gst_pad_get_parent (pad));
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if (!GST_CAPS_IS_FIXED (caps))
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return GST_PAD_CONNECT_DELAYED;
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gst_caps_get_int (caps, "width", &width);
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gst_caps_get_int (caps, "depth", &depth);
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if (width != depth)
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return GST_PAD_CONNECT_REFUSED;
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/* laws 1 and 2 are 1 bps anyway */
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osssink->bps = 1;
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gst_caps_get_int (caps, "law", &law);
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gst_caps_get_int (caps, "endianness", &endianness);
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gst_caps_get_boolean (caps, "signed", &sign);
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if (law == 0) {
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if (width == 16) {
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if (sign == TRUE) {
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if (endianness == G_LITTLE_ENDIAN)
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format = AFMT_S16_LE;
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else if (endianness == G_BIG_ENDIAN)
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format = AFMT_S16_BE;
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}
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else {
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if (endianness == G_LITTLE_ENDIAN)
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format = AFMT_U16_LE;
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else if (endianness == G_BIG_ENDIAN)
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format = AFMT_U16_BE;
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}
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osssink->bps = 2;
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}
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else if (width == 8) {
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if (sign == TRUE) {
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format = AFMT_S8;
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}
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else {
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format = AFMT_U8;
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}
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osssink->bps = 1;
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}
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} else if (law == 1) {
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format = AFMT_MU_LAW;
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} else if (law == 2) {
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format = AFMT_A_LAW;
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} else {
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g_critical ("unknown law");
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return GST_PAD_CONNECT_REFUSED;
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}
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if (format == -1)
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return GST_PAD_CONNECT_REFUSED;
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osssink->format = format;
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gst_caps_get_int (caps, "channels", &osssink->channels);
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gst_caps_get_int (caps, "rate", &osssink->frequency);
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osssink->bps *= osssink->channels;
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osssink->bps *= osssink->frequency;
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if (!gst_osssink_sync_parms (osssink)) {
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return GST_PAD_CONNECT_REFUSED;
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}
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return GST_PAD_CONNECT_OK;
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}
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static gboolean
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gst_osssink_sync_parms (GstOssSink *osssink)
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{
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audio_buf_info ospace;
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int frag;
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gint target_format;
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gint target_channels;
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gint target_frequency;
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GObject *object;
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g_return_val_if_fail (osssink != NULL, FALSE);
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g_return_val_if_fail (GST_IS_OSSSINK (osssink), FALSE);
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if (osssink->fd == -1)
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return FALSE;
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if (osssink->fragment >> 16)
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frag = osssink->fragment;
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else
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frag = 0x7FFF0000 | osssink->fragment;
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GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: trying to set sound card to %dHz %d bit %s (%08x fragment)",
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osssink->frequency, osssink->format,
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(osssink->channels == 2) ? "stereo" : "mono",frag);
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ioctl (osssink->fd, SNDCTL_DSP_SETFRAGMENT, &frag);
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ioctl (osssink->fd, SNDCTL_DSP_RESET, 0);
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target_format = osssink->format;
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target_channels = osssink->channels;
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target_frequency = osssink->frequency;
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ioctl (osssink->fd, SNDCTL_DSP_SETFMT, &osssink->format);
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ioctl (osssink->fd, SNDCTL_DSP_CHANNELS, &osssink->channels);
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ioctl (osssink->fd, SNDCTL_DSP_SPEED, &osssink->frequency);
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ioctl (osssink->fd, SNDCTL_DSP_GETBLKSIZE, &osssink->fragment);
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ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
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GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: set sound card to %dHz %d bit %s (%d bytes buffer, %08x fragment)",
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osssink->frequency, osssink->format,
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(osssink->channels == 2) ? "stereo" : "mono", ospace.bytes, osssink->fragment);
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object = G_OBJECT (osssink);
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g_object_freeze_notify (object);
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g_object_notify (object, "channels");
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g_object_notify (object, "frequency");
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g_object_notify (object, "fragment");
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g_object_notify (object, "format");
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g_object_thaw_notify (object);
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osssink->fragment_time = (1000000 * osssink->fragment) / osssink->bps;
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GST_INFO (GST_CAT_PLUGIN_INFO, "fragment time %u %llu\n", osssink->bps, osssink->fragment_time);
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if (target_format != osssink->format ||
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target_channels != osssink->channels ||
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target_frequency != osssink->frequency)
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{
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g_warning ("could not configure oss with required parameters, enjoy the noise :)");
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/* we could eventually return FALSE here, or just do some additional tests
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* to see that the frequencies don't differ too much etc.. */
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}
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return TRUE;
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}
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static void
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gst_osssink_set_clock (GstElement *element, GstClock *clock)
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{
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GstOssSink *osssink;
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osssink = GST_OSSSINK (element);
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osssink->clock = clock;
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}
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static void
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gst_osssink_chain (GstPad *pad, GstBuffer *buf)
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{
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GstOssSink *osssink;
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GstClockTime buftime;
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/* this has to be an audio buffer */
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osssink = GST_OSSSINK (gst_pad_get_parent (pad));
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if (GST_IS_EVENT (buf)) {
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GstEvent *event = GST_EVENT (buf);
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//gint64 offset;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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ioctl (osssink->fd, SNDCTL_DSP_SYNC);
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gst_pad_event_default (pad, event);
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return;
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case GST_EVENT_NEW_MEDIA:
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g_print ("new media\n");
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return;
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case GST_EVENT_DISCONTINUOUS:
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{
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gint64 value;
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ioctl (osssink->fd, SNDCTL_DSP_RESET);
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if (gst_event_discont_get_value (event, GST_FORMAT_TIME, &value)) {
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gst_clock_handle_discont (osssink->clock, value);
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}
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osssink->resync = TRUE;
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return;
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}
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default:
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gst_pad_event_default (pad, event);
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return;
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}
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gst_event_free (event);
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}
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if (!osssink->bps) {
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gst_buffer_unref (buf);
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gst_element_error (GST_ELEMENT (osssink), "capsnego was never performed, unknown data type");
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return;
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}
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buftime = GST_BUFFER_TIMESTAMP (buf);
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if (osssink->fd >= 0) {
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if (!osssink->mute) {
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guchar *data = GST_BUFFER_DATA (buf);
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gint size = GST_BUFFER_SIZE (buf);
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if (osssink->clock) {
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gint delay;
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gint64 queued;
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GstClockTimeDiff jitter;
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ioctl (osssink->fd, SNDCTL_DSP_GETODELAY, &delay);
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queued = delay * GST_SECOND / osssink->bps;
|
|
|
|
if (osssink->resync) {
|
|
gst_element_clock_wait (GST_ELEMENT (osssink), osssink->clock,
|
|
buftime - queued, &jitter);
|
|
|
|
if (jitter > 0) {
|
|
write (osssink->fd, data, size);
|
|
osssink->resync = FALSE;
|
|
}
|
|
}
|
|
else {
|
|
write (osssink->fd, data, size);
|
|
}
|
|
}
|
|
/* no clock, try to be as fast as possible */
|
|
else {
|
|
audio_buf_info ospace;
|
|
|
|
ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
|
|
|
|
if (ospace.bytes >= size) {
|
|
write (osssink->fd, data, size);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
static void
|
|
gst_osssink_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
|
|
{
|
|
GstOssSink *osssink;
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
g_return_if_fail (GST_IS_OSSSINK (object));
|
|
|
|
osssink = GST_OSSSINK (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DEVICE:
|
|
/* disallow changing the device while it is opened
|
|
get_property("device") should return the right one */
|
|
if (!GST_FLAG_IS_SET (osssink, GST_OSSSINK_OPEN))
|
|
{
|
|
g_free (osssink->device);
|
|
osssink->device = g_strdup (g_value_get_string (value));
|
|
g_object_notify (object, "device");
|
|
}
|
|
break;
|
|
case ARG_MUTE:
|
|
osssink->mute = g_value_get_boolean (value);
|
|
g_object_notify (G_OBJECT (osssink), "mute");
|
|
break;
|
|
case ARG_FORMAT:
|
|
osssink->format = g_value_get_int (value);
|
|
gst_osssink_sync_parms (osssink);
|
|
break;
|
|
case ARG_CHANNELS:
|
|
osssink->channels = g_value_get_enum (value);
|
|
gst_osssink_sync_parms (osssink);
|
|
break;
|
|
case ARG_FREQUENCY:
|
|
osssink->frequency = g_value_get_int (value);
|
|
gst_osssink_sync_parms (osssink);
|
|
break;
|
|
case ARG_FRAGMENT:
|
|
osssink->fragment = g_value_get_int (value);
|
|
gst_osssink_sync_parms (osssink);
|
|
break;
|
|
case ARG_BUFFER_SIZE:
|
|
if (osssink->bufsize == g_value_get_int (value)) break;
|
|
osssink->bufsize = g_value_get_int (value);
|
|
osssink->sinkpool = gst_buffer_pool_get_default (osssink->bufsize, 6);
|
|
g_object_notify (object, "buffer_size");
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_osssink_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
|
|
{
|
|
GstOssSink *osssink;
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
g_return_if_fail (GST_IS_OSSSINK (object));
|
|
|
|
osssink = GST_OSSSINK (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DEVICE:
|
|
g_value_set_string (value, osssink->device);
|
|
break;
|
|
case ARG_MUTE:
|
|
g_value_set_boolean (value, osssink->mute);
|
|
break;
|
|
case ARG_FORMAT:
|
|
g_value_set_int (value, osssink->format);
|
|
break;
|
|
case ARG_CHANNELS:
|
|
g_value_set_enum (value, osssink->channels);
|
|
break;
|
|
case ARG_FREQUENCY:
|
|
g_value_set_int (value, osssink->frequency);
|
|
break;
|
|
case ARG_FRAGMENT:
|
|
g_value_set_int (value, osssink->fragment);
|
|
break;
|
|
case ARG_BUFFER_SIZE:
|
|
g_value_set_int (value, osssink->bufsize);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_osssink_open_audio (GstOssSink *sink)
|
|
{
|
|
gint caps;
|
|
g_return_val_if_fail (sink->fd == -1, FALSE);
|
|
|
|
GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: attempting to open sound device");
|
|
|
|
/* first try to open the sound card */
|
|
sink->fd = open(sink->device, O_WRONLY | O_NONBLOCK);
|
|
if (errno == EBUSY) {
|
|
g_warning ("osssink: unable to open the sound device (in use ?)\n");
|
|
return FALSE;
|
|
}
|
|
|
|
/* re-open the sound device in blocking mode */
|
|
close(sink->fd);
|
|
sink->fd = open(sink->device, O_WRONLY);
|
|
|
|
/* if we have it, set the default parameters and go have fun */
|
|
if (sink->fd >= 0) {
|
|
/* set card state */
|
|
ioctl(sink->fd, SNDCTL_DSP_GETCAPS, &caps);
|
|
|
|
GST_INFO(GST_CAT_PLUGIN_INFO, "osssink: Capabilities %08x", caps);
|
|
|
|
if (caps & DSP_CAP_DUPLEX) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Full duplex");
|
|
if (caps & DSP_CAP_REALTIME) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Realtime");
|
|
if (caps & DSP_CAP_BATCH) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Batch");
|
|
if (caps & DSP_CAP_COPROC) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Has coprocessor");
|
|
if (caps & DSP_CAP_TRIGGER) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Trigger");
|
|
if (caps & DSP_CAP_MMAP) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Direct access");
|
|
|
|
#ifdef DSP_CAP_MULTI
|
|
if (caps & DSP_CAP_MULTI) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Multiple open");
|
|
#endif /* DSP_CAP_MULTI */
|
|
|
|
#ifdef DSP_CAP_BIND
|
|
if (caps & DSP_CAP_BIND) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Channel binding");
|
|
#endif /* DSP_CAP_BIND */
|
|
|
|
ioctl(sink->fd, SNDCTL_DSP_GETFMTS, &caps);
|
|
|
|
GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Formats %08x", caps);
|
|
if (caps & AFMT_MU_LAW) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: MU_LAW");
|
|
if (caps & AFMT_A_LAW) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: A_LAW");
|
|
if (caps & AFMT_IMA_ADPCM) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: IMA_ADPCM");
|
|
if (caps & AFMT_U8) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: U8");
|
|
if (caps & AFMT_S16_LE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: S16_LE");
|
|
if (caps & AFMT_S16_BE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: S16_BE");
|
|
if (caps & AFMT_S8) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: S8");
|
|
if (caps & AFMT_U16_LE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: U16_LE");
|
|
if (caps & AFMT_U16_BE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: U16_BE");
|
|
if (caps & AFMT_MPEG) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: MPEG");
|
|
#ifdef AFMT_AC3
|
|
if (caps & AFMT_AC3) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: AC3");
|
|
#endif
|
|
|
|
GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: opened audio (%s) with fd=%d", sink->device, sink->fd);
|
|
GST_FLAG_SET (sink, GST_OSSSINK_OPEN);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
gst_osssink_close_audio (GstOssSink *sink)
|
|
{
|
|
if (sink->fd < 0) return;
|
|
|
|
close(sink->fd);
|
|
sink->fd = -1;
|
|
|
|
GST_FLAG_UNSET (sink, GST_OSSSINK_OPEN);
|
|
|
|
GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: closed sound device");
|
|
}
|
|
|
|
static GstElementStateReturn
|
|
gst_osssink_change_state (GstElement *element)
|
|
{
|
|
GstOssSink *osssink;
|
|
|
|
g_return_val_if_fail (GST_IS_OSSSINK (element), FALSE);
|
|
|
|
osssink = GST_OSSSINK (element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_NULL_TO_READY:
|
|
if (!GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN)) {
|
|
if (!gst_osssink_open_audio (osssink)) {
|
|
return GST_STATE_FAILURE;
|
|
}
|
|
}
|
|
break;
|
|
case GST_STATE_READY_TO_PAUSED:
|
|
case GST_STATE_PAUSED_TO_PLAYING:
|
|
osssink->resync = TRUE;
|
|
break;
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
{
|
|
if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
|
|
ioctl (osssink->fd, SNDCTL_DSP_RESET, 0);
|
|
break;
|
|
}
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
|
|
ioctl (osssink->fd, SNDCTL_DSP_RESET, 0);
|
|
break;
|
|
case GST_STATE_READY_TO_NULL:
|
|
if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
|
|
gst_osssink_close_audio (osssink);
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
|
|
return GST_STATE_SUCCESS;
|
|
}
|
|
|
|
gboolean
|
|
gst_osssink_factory_init (GstPlugin *plugin)
|
|
{
|
|
GstElementFactory *factory;
|
|
|
|
factory = gst_element_factory_new ("osssink", GST_TYPE_OSSSINK, &gst_osssink_details);
|
|
g_return_val_if_fail (factory != NULL, FALSE);
|
|
|
|
gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (osssink_sink_factory));
|
|
|
|
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
|
|
|
|
return TRUE;
|
|
}
|
|
|