gstreamer/gst/elements/gstaudiosink.c

/* GStreamer
 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
 *                    2000 Wim Taymans <wtay@chello.be>
 *
 * 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 <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#include <unistd.h>
#include <errno.h>

//#define DEBUG_ENABLED

#include <gstaudiosink.h>
#include <gst/meta/audioraw.h>


GstElementDetails gst_audiosink_details = {  
  "Audio Sink (OSS)",
  "Sink/Audio",
  "Output to a sound card via OSS",
  VERSION,
  "Erik Walthinsen <omega@cse.ogi.edu>",
  "(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;
}