mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1e04cefaf2
Original commit message from CVS: Mega update of INFO, DEBUG, and ERROR subsystems, renamed with GST_ prefix. GST_DEBUG now takes a category parameter, which is the same as GST_INFO system. They are now called GST_CAT_*. All the GST_DEBUGs are set to 0 for now, we need to go and fix all these eventually.
429 lines
12 KiB
C
429 lines
12 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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*
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* gstaudiosink.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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//#define DEBUG_ENABLED
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#include <gstaudiosink.h>
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#include <gst/meta/audioraw.h>
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GstElementDetails gst_audiosink_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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"(C) 1999",
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};
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static void gst_audiosink_class_init (GstAudioSinkClass *klass);
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static void gst_audiosink_init (GstAudioSink *audiosink);
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static gboolean gst_audiosink_open_audio (GstAudioSink *sink);
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static void gst_audiosink_close_audio (GstAudioSink *sink);
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static GstElementStateReturn gst_audiosink_change_state (GstElement *element);
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static void gst_audiosink_set_arg (GtkObject *object, GtkArg *arg, guint id);
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static void gst_audiosink_get_arg (GtkObject *object, GtkArg *arg, guint id);
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static void gst_audiosink_chain (GstPad *pad,GstBuffer *buf);
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/* AudioSink 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_MUTE,
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ARG_FORMAT,
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ARG_CHANNELS,
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ARG_FREQUENCY,
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/* FILL ME */
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};
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static GstPadFactory audiosink_sink_factory = {
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"sink",
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GST_PAD_FACTORY_SINK,
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GST_PAD_FACTORY_ALWAYS,
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GST_PAD_FACTORY_CAPS (
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"audiosink_sink",
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"audio/raw",
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"format", GST_PROPS_INT (AFMT_S16_LE),
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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 (8000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2)
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),
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NULL
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};
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#define GST_TYPE_AUDIOSINK_FORMATS (gst_audiosink_formats_get_type())
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static GtkType
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gst_audiosink_formats_get_type(void) {
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static GtkType audiosink_formats_type = 0;
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static GtkEnumValue audiosink_formats[] = {
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{8, "8", "8 Bits"},
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{16, "16", "16 Bits"},
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{0, NULL, NULL},
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};
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if (!audiosink_formats_type) {
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audiosink_formats_type = gtk_type_register_enum("GstAudiosinkFormats", audiosink_formats);
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}
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return audiosink_formats_type;
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}
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#define GST_TYPE_AUDIOSINK_CHANNELS (gst_audiosink_channels_get_type())
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static GtkType
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gst_audiosink_channels_get_type(void) {
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static GtkType audiosink_channels_type = 0;
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static GtkEnumValue audiosink_channels[] = {
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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 (!audiosink_channels_type) {
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audiosink_channels_type = gtk_type_register_enum("GstAudiosinkChannels", audiosink_channels);
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}
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return audiosink_channels_type;
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}
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static GstElementClass *parent_class = NULL;
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static guint gst_audiosink_signals[LAST_SIGNAL] = { 0 };
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static GstPadTemplate *gst_audiosink_sink_template;
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GtkType
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gst_audiosink_get_type (void)
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{
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static GtkType audiosink_type = 0;
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if (!audiosink_type) {
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static const GtkTypeInfo audiosink_info = {
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"GstAudioSink",
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sizeof(GstAudioSink),
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sizeof(GstAudioSinkClass),
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(GtkClassInitFunc)gst_audiosink_class_init,
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(GtkObjectInitFunc)gst_audiosink_init,
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(GtkArgSetFunc)NULL,
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(GtkArgGetFunc)NULL,
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(GtkClassInitFunc)NULL,
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};
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audiosink_type = gtk_type_unique (GST_TYPE_ELEMENT, &audiosink_info);
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}
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return audiosink_type;
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}
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static void
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gst_audiosink_class_init (GstAudioSinkClass *klass)
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{
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GtkObjectClass *gtkobject_class;
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GstElementClass *gstelement_class;
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gtkobject_class = (GtkObjectClass*)klass;
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gstelement_class = (GstElementClass*)klass;
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parent_class = gtk_type_class(GST_TYPE_ELEMENT);
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gtk_object_add_arg_type ("GstAudioSink::mute", GTK_TYPE_BOOL,
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GTK_ARG_READWRITE, ARG_MUTE);
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gtk_object_add_arg_type ("GstAudioSink::format", GST_TYPE_AUDIOSINK_FORMATS,
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GTK_ARG_READWRITE, ARG_FORMAT);
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gtk_object_add_arg_type ("GstAudioSink::channels", GST_TYPE_AUDIOSINK_CHANNELS,
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GTK_ARG_READWRITE, ARG_CHANNELS);
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gtk_object_add_arg_type ("GstAudioSink::frequency", GTK_TYPE_INT,
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GTK_ARG_READWRITE, ARG_FREQUENCY);
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gtkobject_class->set_arg = gst_audiosink_set_arg;
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gtkobject_class->get_arg = gst_audiosink_get_arg;
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gst_audiosink_signals[SIGNAL_HANDOFF] =
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gtk_signal_new("handoff",GTK_RUN_LAST,gtkobject_class->type,
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GTK_SIGNAL_OFFSET(GstAudioSinkClass,handoff),
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gtk_marshal_NONE__NONE,GTK_TYPE_NONE,0);
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gtk_object_class_add_signals(gtkobject_class,gst_audiosink_signals,
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LAST_SIGNAL);
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gstelement_class->change_state = gst_audiosink_change_state;
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}
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static void
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gst_audiosink_init (GstAudioSink *audiosink)
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{
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audiosink->sinkpad = gst_pad_new_from_template (gst_audiosink_sink_template, "sink");
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gst_element_add_pad (GST_ELEMENT (audiosink), audiosink->sinkpad);
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gst_pad_set_chain_function (audiosink->sinkpad, gst_audiosink_chain);
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audiosink->fd = -1;
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audiosink->clock = gst_clock_get_system();
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audiosink->format = 16;
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audiosink->channels = 2;
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audiosink->frequency = 44100;
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gst_clock_register (audiosink->clock, GST_OBJECT (audiosink));
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GST_FLAG_SET (audiosink, GST_ELEMENT_THREAD_SUGGESTED);
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}
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static void
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gst_audiosink_sync_parms (GstAudioSink *audiosink)
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{
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audio_buf_info ospace;
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int frag;
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g_return_if_fail (audiosink != NULL);
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g_return_if_fail (GST_IS_AUDIOSINK (audiosink));
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if (audiosink->fd == -1) return;
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ioctl (audiosink->fd,SNDCTL_DSP_RESET, 0);
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ioctl (audiosink->fd, SNDCTL_DSP_SETFMT, &audiosink->format);
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ioctl (audiosink->fd, SNDCTL_DSP_CHANNELS, &audiosink->channels);
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ioctl (audiosink->fd, SNDCTL_DSP_SPEED, &audiosink->frequency);
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ioctl (audiosink->fd, SNDCTL_DSP_GETBLKSIZE, &frag);
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ioctl (audiosink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
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g_print("audiosink: setting sound card to %dKHz %d bit %s (%d bytes buffer, %d fragment)\n",
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audiosink->frequency, audiosink->format,
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(audiosink->channels == 2) ? "stereo" : "mono", ospace.bytes, frag);
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}
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static void
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gst_audiosink_chain (GstPad *pad, GstBuffer *buf)
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{
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GstAudioSink *audiosink;
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MetaAudioRaw *meta;
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gboolean in_flush;
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audio_buf_info ospace;
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g_return_if_fail (pad != NULL);
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (buf != NULL);
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/* this has to be an audio buffer */
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// g_return_if_fail(((GstMeta *)buf->meta)->type !=
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//gst_audiosink_type_audio);
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audiosink = GST_AUDIOSINK (pad->parent);
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// g_return_if_fail(GST_FLAG_IS_SET(audiosink,GST_STATE_RUNNING));
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if ((in_flush = GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLUSH))) {
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GST_DEBUG (0,"audiosink: flush\n");
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ioctl (audiosink->fd, SNDCTL_DSP_RESET, 0);
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}
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meta = (MetaAudioRaw *)gst_buffer_get_first_meta (buf);
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if (meta != NULL) {
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if ((meta->format != audiosink->format) ||
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(meta->channels != audiosink->channels) ||
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(meta->frequency != audiosink->frequency))
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{
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audiosink->format = meta->format;
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audiosink->channels = meta->channels;
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audiosink->frequency = meta->frequency;
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gst_audiosink_sync_parms (audiosink);
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g_print("audiosink: sound device set to format %d, %d channels, %dHz\n",
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audiosink->format, audiosink->channels, audiosink->frequency);
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}
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}
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gtk_signal_emit (GTK_OBJECT (audiosink), gst_audiosink_signals[SIGNAL_HANDOFF],
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audiosink);
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if (GST_BUFFER_DATA (buf) != NULL) {
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gst_trace_add_entry(NULL, 0, buf, "audiosink: writing to soundcard");
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//g_print("audiosink: writing to soundcard\n");
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if (audiosink->fd > 2) {
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if (!audiosink->mute) {
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gst_clock_wait (audiosink->clock, GST_BUFFER_TIMESTAMP (buf), GST_OBJECT (audiosink));
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ioctl (audiosink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
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GST_DEBUG (0,"audiosink: (%d bytes buffer) %d %p %d\n", ospace.bytes,
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audiosink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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write (audiosink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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//write(STDOUT_FILENO,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf));
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}
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}
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}
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gst_buffer_unref (buf);
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}
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static void
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gst_audiosink_set_arg (GtkObject *object, GtkArg *arg, guint id)
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{
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GstAudioSink *audiosink;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail (GST_IS_AUDIOSINK (object));
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audiosink = GST_AUDIOSINK (object);
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switch(id) {
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case ARG_MUTE:
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audiosink->mute = GTK_VALUE_BOOL (*arg);
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break;
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case ARG_FORMAT:
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audiosink->format = GTK_VALUE_ENUM (*arg);
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gst_audiosink_sync_parms (audiosink);
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break;
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case ARG_CHANNELS:
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audiosink->channels = GTK_VALUE_ENUM (*arg);
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gst_audiosink_sync_parms (audiosink);
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break;
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case ARG_FREQUENCY:
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audiosink->frequency = GTK_VALUE_INT (*arg);
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gst_audiosink_sync_parms (audiosink);
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break;
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default:
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break;
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}
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}
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static void
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gst_audiosink_get_arg (GtkObject *object, GtkArg *arg, guint id)
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{
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GstAudioSink *audiosink;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail (GST_IS_AUDIOSINK (object));
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audiosink = GST_AUDIOSINK (object);
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switch(id) {
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case ARG_MUTE:
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GTK_VALUE_BOOL (*arg) = audiosink->mute;
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break;
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case ARG_FORMAT:
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GTK_VALUE_ENUM (*arg) = audiosink->format;
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break;
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case ARG_CHANNELS:
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GTK_VALUE_ENUM (*arg) = audiosink->channels;
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break;
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case ARG_FREQUENCY:
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GTK_VALUE_INT (*arg) = audiosink->frequency;
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break;
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default:
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break;
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}
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}
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static gboolean
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gst_audiosink_open_audio (GstAudioSink *sink)
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{
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g_return_val_if_fail (sink->fd == -1, FALSE);
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g_print ("audiosink: attempting to open sound device\n");
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/* first try to open the sound card */
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sink->fd = open("/dev/dsp", O_WRONLY);
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/* if we have it, set the default parameters and go have fun */
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if (sink->fd > 0) {
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/* set card state */
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sink->format = AFMT_S16_LE;
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sink->channels = 2; /* stereo */
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sink->frequency = 44100;
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gst_audiosink_sync_parms (sink);
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ioctl(sink->fd, SNDCTL_DSP_GETCAPS, &sink->caps);
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g_print("audiosink: Capabilities\n");
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if (sink->caps & DSP_CAP_DUPLEX) g_print("audiosink: Full duplex\n");
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if (sink->caps & DSP_CAP_REALTIME) g_print("audiosink: Realtime\n");
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if (sink->caps & DSP_CAP_BATCH) g_print("audiosink: Batch\n");
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if (sink->caps & DSP_CAP_COPROC) g_print("audiosink: Has coprocessor\n");
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if (sink->caps & DSP_CAP_TRIGGER) g_print("audiosink: Trigger\n");
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if (sink->caps & DSP_CAP_MMAP) g_print("audiosink: Direct access\n");
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g_print("audiosink: opened audio with fd=%d\n", sink->fd);
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GST_FLAG_SET (sink, GST_AUDIOSINK_OPEN);
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return TRUE;
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}
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return FALSE;
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}
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static void
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gst_audiosink_close_audio (GstAudioSink *sink)
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{
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if (sink->fd < 0) return;
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close(sink->fd);
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sink->fd = -1;
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GST_FLAG_UNSET (sink, GST_AUDIOSINK_OPEN);
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g_print("audiosink: closed sound device\n");
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}
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static GstElementStateReturn
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gst_audiosink_change_state (GstElement *element)
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{
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g_return_val_if_fail (GST_IS_AUDIOSINK (element), FALSE);
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/* if going down into NULL state, close the file if it's open */
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if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
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if (GST_FLAG_IS_SET (element, GST_AUDIOSINK_OPEN))
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gst_audiosink_close_audio (GST_AUDIOSINK (element));
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/* otherwise (READY or higher) we need to open the sound card */
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} else {
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if (!GST_FLAG_IS_SET (element, GST_AUDIOSINK_OPEN)) {
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if (!gst_audiosink_open_audio (GST_AUDIOSINK (element)))
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return GST_STATE_FAILURE;
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}
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}
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if (GST_ELEMENT_CLASS (parent_class)->change_state)
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return GST_ELEMENT_CLASS (parent_class)->change_state (element);
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return GST_STATE_SUCCESS;
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}
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gboolean
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gst_audiosink_factory_init (GstElementFactory *factory)
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{
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gst_audiosink_sink_template = gst_padtemplate_new (&audiosink_sink_factory);
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gst_elementfactory_add_padtemplate (factory, gst_audiosink_sink_template);
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return TRUE;
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}
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