/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000 Wim Taymans * * gstaudiosink.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. */ #include #include #include #include #include #include #include //#define DEBUG_ENABLED #include #include GstElementDetails gst_audiosink_details = { "Audio Sink (OSS)", "Sink/Audio", "Output to a sound card via OSS", VERSION, "Erik Walthinsen ", "(C) 1999", }; static void gst_audiosink_class_init (GstAudioSinkClass *klass); static void gst_audiosink_init (GstAudioSink *audiosink); static gboolean gst_audiosink_open_audio (GstAudioSink *sink); static void gst_audiosink_close_audio (GstAudioSink *sink); static GstElementStateReturn gst_audiosink_change_state (GstElement *element); static void gst_audiosink_set_arg (GtkObject *object, GtkArg *arg, guint id); static void gst_audiosink_get_arg (GtkObject *object, GtkArg *arg, guint id); static void gst_audiosink_chain (GstPad *pad,GstBuffer *buf); /* AudioSink signals and args */ enum { SIGNAL_HANDOFF, LAST_SIGNAL }; enum { ARG_0, ARG_MUTE, ARG_FORMAT, ARG_CHANNELS, ARG_FREQUENCY, /* FILL ME */ }; static GstPadFactory audiosink_sink_factory = { "sink", GST_PAD_FACTORY_SINK, GST_PAD_FACTORY_ALWAYS, GST_PAD_FACTORY_CAPS ( "audiosink_sink", "audio/raw", "format", GST_PROPS_INT (AFMT_S16_LE), "depth", GST_PROPS_LIST ( GST_PROPS_INT (8), GST_PROPS_INT (16) ), "rate", GST_PROPS_INT_RANGE (8000, 48000), "channels", GST_PROPS_INT_RANGE (1, 2) ), NULL }; #define GST_TYPE_AUDIOSINK_FORMATS (gst_audiosink_formats_get_type()) static GtkType gst_audiosink_formats_get_type(void) { static GtkType audiosink_formats_type = 0; static GtkEnumValue audiosink_formats[] = { {8, "8", "8 Bits"}, {16, "16", "16 Bits"}, {0, NULL, NULL}, }; if (!audiosink_formats_type) { audiosink_formats_type = gtk_type_register_enum("GstAudiosinkFormats", audiosink_formats); } return audiosink_formats_type; } #define GST_TYPE_AUDIOSINK_CHANNELS (gst_audiosink_channels_get_type()) static GtkType gst_audiosink_channels_get_type(void) { static GtkType audiosink_channels_type = 0; static GtkEnumValue audiosink_channels[] = { {1, "1", "Mono"}, {2, "2", "Stereo"}, {0, NULL, NULL}, }; if (!audiosink_channels_type) { audiosink_channels_type = gtk_type_register_enum("GstAudiosinkChannels", audiosink_channels); } return audiosink_channels_type; } static GstElementClass *parent_class = NULL; static guint gst_audiosink_signals[LAST_SIGNAL] = { 0 }; static GstPadTemplate *gst_audiosink_sink_template; GtkType gst_audiosink_get_type (void) { static GtkType audiosink_type = 0; if (!audiosink_type) { static const GtkTypeInfo audiosink_info = { "GstAudioSink", sizeof(GstAudioSink), sizeof(GstAudioSinkClass), (GtkClassInitFunc)gst_audiosink_class_init, (GtkObjectInitFunc)gst_audiosink_init, (GtkArgSetFunc)NULL, (GtkArgGetFunc)NULL, (GtkClassInitFunc)NULL, }; audiosink_type = gtk_type_unique (GST_TYPE_ELEMENT, &audiosink_info); } return audiosink_type; } static void gst_audiosink_class_init (GstAudioSinkClass *klass) { GtkObjectClass *gtkobject_class; GstElementClass *gstelement_class; gtkobject_class = (GtkObjectClass*)klass; gstelement_class = (GstElementClass*)klass; parent_class = gtk_type_class(GST_TYPE_ELEMENT); gtk_object_add_arg_type ("GstAudioSink::mute", GTK_TYPE_BOOL, GTK_ARG_READWRITE, ARG_MUTE); gtk_object_add_arg_type ("GstAudioSink::format", GST_TYPE_AUDIOSINK_FORMATS, GTK_ARG_READWRITE, ARG_FORMAT); gtk_object_add_arg_type ("GstAudioSink::channels", GST_TYPE_AUDIOSINK_CHANNELS, GTK_ARG_READWRITE, ARG_CHANNELS); gtk_object_add_arg_type ("GstAudioSink::frequency", GTK_TYPE_INT, GTK_ARG_READWRITE, ARG_FREQUENCY); gtkobject_class->set_arg = gst_audiosink_set_arg; gtkobject_class->get_arg = gst_audiosink_get_arg; gst_audiosink_signals[SIGNAL_HANDOFF] = gtk_signal_new("handoff",GTK_RUN_LAST,gtkobject_class->type, GTK_SIGNAL_OFFSET(GstAudioSinkClass,handoff), gtk_marshal_NONE__NONE,GTK_TYPE_NONE,0); gtk_object_class_add_signals(gtkobject_class,gst_audiosink_signals, LAST_SIGNAL); gstelement_class->change_state = gst_audiosink_change_state; } static void gst_audiosink_init (GstAudioSink *audiosink) { audiosink->sinkpad = gst_pad_new_from_template (gst_audiosink_sink_template, "sink"); gst_element_add_pad (GST_ELEMENT (audiosink), audiosink->sinkpad); gst_pad_set_chain_function (audiosink->sinkpad, gst_audiosink_chain); audiosink->fd = -1; audiosink->clock = gst_clock_get_system(); audiosink->format = 16; audiosink->channels = 2; audiosink->frequency = 44100; gst_clock_register (audiosink->clock, GST_OBJECT (audiosink)); GST_FLAG_SET (audiosink, GST_ELEMENT_THREAD_SUGGESTED); } static void gst_audiosink_sync_parms (GstAudioSink *audiosink) { audio_buf_info ospace; int frag; g_return_if_fail (audiosink != NULL); g_return_if_fail (GST_IS_AUDIOSINK (audiosink)); if (audiosink->fd == -1) return; ioctl (audiosink->fd, SNDCTL_DSP_RESET, 0); ioctl (audiosink->fd, SNDCTL_DSP_SETFMT, &audiosink->format); ioctl (audiosink->fd, SNDCTL_DSP_CHANNELS, &audiosink->channels); ioctl (audiosink->fd, SNDCTL_DSP_SPEED, &audiosink->frequency); ioctl (audiosink->fd, SNDCTL_DSP_GETBLKSIZE, &frag); ioctl (audiosink->fd, SNDCTL_DSP_GETOSPACE, &ospace); g_print("audiosink: setting sound card to %dKHz %d bit %s (%d bytes buffer, %d fragment)\n", audiosink->frequency, audiosink->format, (audiosink->channels == 2) ? "stereo" : "mono", ospace.bytes, frag); } static void gst_audiosink_chain (GstPad *pad, GstBuffer *buf) { GstAudioSink *audiosink; MetaAudioRaw *meta; gboolean in_flush; audio_buf_info ospace; g_return_if_fail (pad != NULL); g_return_if_fail (GST_IS_PAD (pad)); g_return_if_fail (buf != NULL); /* this has to be an audio buffer */ // g_return_if_fail(((GstMeta *)buf->meta)->type != //gst_audiosink_type_audio); audiosink = GST_AUDIOSINK (gst_pad_get_parent (pad)); // g_return_if_fail(GST_FLAG_IS_SET(audiosink,GST_STATE_RUNNING)); if ((in_flush = GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLUSH))) { GST_DEBUG (0,"audiosink: flush\n"); ioctl (audiosink->fd, SNDCTL_DSP_RESET, 0); } meta = (MetaAudioRaw *)gst_buffer_get_first_meta (buf); if (meta != NULL) { if ((meta->format != audiosink->format) || (meta->channels != audiosink->channels) || (meta->frequency != audiosink->frequency)) { audiosink->format = meta->format; audiosink->channels = meta->channels; audiosink->frequency = meta->frequency; gst_audiosink_sync_parms (audiosink); g_print("audiosink: sound device set to format %d, %d channels, %dHz\n", audiosink->format, audiosink->channels, audiosink->frequency); } } gtk_signal_emit (GTK_OBJECT (audiosink), gst_audiosink_signals[SIGNAL_HANDOFF], audiosink); if (GST_BUFFER_DATA (buf) != NULL) { gst_trace_add_entry(NULL, 0, buf, "audiosink: writing to soundcard"); //g_print("audiosink: writing to soundcard\n"); if (audiosink->fd >= 0) { if (!audiosink->mute) { gst_clock_wait (audiosink->clock, GST_BUFFER_TIMESTAMP (buf), GST_OBJECT (audiosink)); ioctl (audiosink->fd, SNDCTL_DSP_GETOSPACE, &ospace); GST_DEBUG (0,"audiosink: (%d bytes buffer) %d %p %d\n", ospace.bytes, audiosink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); write (audiosink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); //write(STDOUT_FILENO,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf)); } } } gst_buffer_unref (buf); } static void gst_audiosink_set_arg (GtkObject *object, GtkArg *arg, guint id) { GstAudioSink *audiosink; /* it's not null if we got it, but it might not be ours */ g_return_if_fail (GST_IS_AUDIOSINK (object)); audiosink = GST_AUDIOSINK (object); switch(id) { case ARG_MUTE: audiosink->mute = GTK_VALUE_BOOL (*arg); break; case ARG_FORMAT: audiosink->format = GTK_VALUE_ENUM (*arg); gst_audiosink_sync_parms (audiosink); break; case ARG_CHANNELS: audiosink->channels = GTK_VALUE_ENUM (*arg); gst_audiosink_sync_parms (audiosink); break; case ARG_FREQUENCY: audiosink->frequency = GTK_VALUE_INT (*arg); gst_audiosink_sync_parms (audiosink); break; default: break; } } static void gst_audiosink_get_arg (GtkObject *object, GtkArg *arg, guint id) { GstAudioSink *audiosink; /* it's not null if we got it, but it might not be ours */ g_return_if_fail (GST_IS_AUDIOSINK (object)); audiosink = GST_AUDIOSINK (object); switch(id) { case ARG_MUTE: GTK_VALUE_BOOL (*arg) = audiosink->mute; break; case ARG_FORMAT: GTK_VALUE_ENUM (*arg) = audiosink->format; break; case ARG_CHANNELS: GTK_VALUE_ENUM (*arg) = audiosink->channels; break; case ARG_FREQUENCY: GTK_VALUE_INT (*arg) = audiosink->frequency; break; default: break; } } static gboolean gst_audiosink_open_audio (GstAudioSink *sink) { g_return_val_if_fail (sink->fd == -1, FALSE); g_print ("audiosink: attempting to open sound device\n"); /* first try to open the sound card */ sink->fd = open("/dev/dsp", O_WRONLY | O_NONBLOCK); if (errno == EBUSY) { g_print ("audiosink: 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("/dev/dsp", O_WRONLY); /* if we have it, set the default parameters and go have fun */ if (sink->fd >= 0) { /* set card state */ sink->format = AFMT_S16_LE; sink->channels = 2; /* stereo */ sink->frequency = 44100; gst_audiosink_sync_parms (sink); ioctl(sink->fd, SNDCTL_DSP_GETCAPS, &sink->caps); g_print("audiosink: Capabilities\n"); if (sink->caps & DSP_CAP_DUPLEX) g_print("audiosink: Full duplex\n"); if (sink->caps & DSP_CAP_REALTIME) g_print("audiosink: Realtime\n"); if (sink->caps & DSP_CAP_BATCH) g_print("audiosink: Batch\n"); if (sink->caps & DSP_CAP_COPROC) g_print("audiosink: Has coprocessor\n"); if (sink->caps & DSP_CAP_TRIGGER) g_print("audiosink: Trigger\n"); if (sink->caps & DSP_CAP_MMAP) g_print("audiosink: Direct access\n"); g_print("audiosink: opened audio with fd=%d\n", sink->fd); GST_FLAG_SET (sink, GST_AUDIOSINK_OPEN); return TRUE; } return FALSE; } static void gst_audiosink_close_audio (GstAudioSink *sink) { if (sink->fd < 0) return; close(sink->fd); sink->fd = -1; GST_FLAG_UNSET (sink, GST_AUDIOSINK_OPEN); g_print("audiosink: closed sound device\n"); } static GstElementStateReturn gst_audiosink_change_state (GstElement *element) { g_return_val_if_fail (GST_IS_AUDIOSINK (element), FALSE); /* if going down into NULL state, close the file if it's open */ if (GST_STATE_PENDING (element) == GST_STATE_NULL) { if (GST_FLAG_IS_SET (element, GST_AUDIOSINK_OPEN)) gst_audiosink_close_audio (GST_AUDIOSINK (element)); /* otherwise (READY or higher) we need to open the sound card */ } else { if (!GST_FLAG_IS_SET (element, GST_AUDIOSINK_OPEN)) { if (!gst_audiosink_open_audio (GST_AUDIOSINK (element))) { //return GST_STATE_FAILURE; } } } if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element); return GST_STATE_SUCCESS; } gboolean gst_audiosink_factory_init (GstElementFactory *factory) { gst_audiosink_sink_template = gst_padtemplate_new (&audiosink_sink_factory); gst_elementfactory_add_padtemplate (factory, gst_audiosink_sink_template); return TRUE; }