gstreamer/gst/wavparse/gstwavparse.c

/* GStreamer
 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
 *
 * 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 <string.h>

#include <gstwavparse.h>

static void		gst_wavparse_class_init		(GstWavParseClass *klass);
static void		gst_wavparse_init		(GstWavParse *wavparse);

static GstCaps*		wav_type_find			(GstBuffer *buf, gpointer private);

static const GstFormat*	gst_wavparse_get_formats	(GstPad *pad);
static const GstQueryType *
			gst_wavparse_get_query_types	(GstPad *pad);
static gboolean		gst_wavparse_pad_query		(GstPad *pad, 
		                                	 GstQueryType type,
							 GstFormat *format, 
							 gint64 *value);
static gboolean		gst_wavparse_pad_convert 	(GstPad *pad,
							 GstFormat src_format,
							 gint64 src_value,
							 GstFormat *dest_format,
							 gint64 *dest_value);
static void		gst_wavparse_chain		(GstPad *pad, GstBuffer *buf);

static const GstEventMask*
			gst_wavparse_get_event_masks 	(GstPad *pad);
static gboolean 	gst_wavparse_srcpad_event 	(GstPad *pad, GstEvent *event);

/* elementfactory information */
static GstElementDetails gst_wavparse_details = {
  ".wav parser",
  "Codec/Parser",
  "LGPL",
  "Parse a .wav file into raw audio",
  VERSION,
  "Erik Walthinsen <omega@cse.ogi.edu>",
  "(C) 1999",
};

GST_PAD_TEMPLATE_FACTORY (sink_template_factory,
  "wavparse_sink",
  GST_PAD_SINK,
  GST_PAD_ALWAYS,
  GST_CAPS_NEW (
    "wavparse_wav",
    "audio/x-wav",
    NULL
  )
)

GST_PAD_TEMPLATE_FACTORY (src_template_factory,
  "wavparse_src",
  GST_PAD_SRC,
  GST_PAD_ALWAYS,
  GST_CAPS_NEW (
    "wavparse_raw",
    "audio/raw",
      "format",            GST_PROPS_STRING ("int"),
       "law",              GST_PROPS_INT (0),
       "endianness",       GST_PROPS_INT (G_LITTLE_ENDIAN),
       "signed",           GST_PROPS_LIST (
				GST_PROPS_BOOLEAN (FALSE),
				GST_PROPS_BOOLEAN (TRUE)
			   ),
       "width",            GST_PROPS_LIST (
	                     GST_PROPS_INT (8),
	                     GST_PROPS_INT (16)
			   ),
       "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)
  ),
  GST_CAPS_NEW (
    "wavparse_mp3",
    "audio/x-mp3",
    NULL
  ),
  GST_CAPS_NEW (
    "parsewav_law",
    "audio/raw",
      "format",            GST_PROPS_STRING ("int"),
      "law",               GST_PROPS_INT_RANGE (1, 2),
      "endianness",        GST_PROPS_INT (G_LITTLE_ENDIAN),
      "width",             GST_PROPS_INT (8),
      "depth",             GST_PROPS_INT (8),
      "rate",              GST_PROPS_INT_RANGE (8000, 48000),
      "channels",          GST_PROPS_INT_RANGE (1, 2)
  )
)

/* typefactory for 'wav' */
static GstTypeDefinition
wavdefinition =
{
  "wavparse_audio/x-wav",
  "audio/x-wav",
  ".wav",
  wav_type_find,
};


/* WavParse signals and args */
enum {
  /* FILL ME */
  LAST_SIGNAL
};

enum {
  ARG_0,
  /* FILL ME */
};

static GstElementClass *parent_class = NULL;
/*static guint gst_wavparse_signals[LAST_SIGNAL] = { 0 }; */

GType
gst_wavparse_get_type (void) 
{
  static GType wavparse_type = 0;

  if (!wavparse_type) {
    static const GTypeInfo wavparse_info = {
      sizeof(GstWavParseClass),      NULL,
      NULL,
      (GClassInitFunc) gst_wavparse_class_init,
      NULL,
      NULL,
      sizeof(GstWavParse),
      0,
      (GInstanceInitFunc) gst_wavparse_init,
    };
    wavparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse", &wavparse_info, 0);
  }
  return wavparse_type;
}

static void
gst_wavparse_class_init (GstWavParseClass *klass) 
{
  GstElementClass *gstelement_class;

  gstelement_class = (GstElementClass*) klass;

  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
}

static void 
gst_wavparse_init (GstWavParse *wavparse) 
{
  /* sink */
  wavparse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_template_factory), "sink");
  gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);

  gst_pad_set_formats_function (wavparse->sinkpad, gst_wavparse_get_formats);
  gst_pad_set_convert_function (wavparse->sinkpad, gst_wavparse_pad_convert);
  gst_pad_set_query_type_function (wavparse->sinkpad, 
		                   gst_wavparse_get_query_types);
  gst_pad_set_query_function (wavparse->sinkpad, gst_wavparse_pad_query);

  /* source */
  wavparse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_template_factory), "src");
  gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
  gst_pad_set_formats_function (wavparse->srcpad, gst_wavparse_get_formats);
  gst_pad_set_convert_function (wavparse->srcpad, gst_wavparse_pad_convert);
  gst_pad_set_query_type_function (wavparse->srcpad,
		                   gst_wavparse_get_query_types);
  gst_pad_set_query_function (wavparse->srcpad, gst_wavparse_pad_query);
  gst_pad_set_event_function (wavparse->srcpad, gst_wavparse_srcpad_event);
  gst_pad_set_event_mask_function (wavparse->srcpad, gst_wavparse_get_event_masks);

  gst_pad_set_chain_function (wavparse->sinkpad, gst_wavparse_chain);


  wavparse->riff = NULL;

  wavparse->state = GST_WAVPARSE_UNKNOWN;
  wavparse->riff = NULL;
  wavparse->riff_nextlikely = 0;
  wavparse->size = 0;
  wavparse->bps = 0;
  wavparse->offset = 0;
  wavparse->need_discont = FALSE;
}

static GstCaps*
wav_type_find (GstBuffer *buf, gpointer private)
{
  gchar *data = GST_BUFFER_DATA (buf);

  if (GST_BUFFER_SIZE (buf) < 12) return NULL;
  if (strncmp (&data[0], "RIFF", 4)) return NULL;
  if (strncmp (&data[8], "WAVE", 4)) return NULL;

  return gst_caps_new ("wav_type_find", "audio/x-wav", NULL);
}

static void
gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
{
  GstWavParse *wavparse;
  gboolean buffer_riffed = FALSE;	/* so we don't parse twice */
  gulong size;

  g_return_if_fail (pad != NULL);
  g_return_if_fail (GST_IS_PAD (pad));
  g_return_if_fail (buf != NULL);
  g_return_if_fail (GST_BUFFER_DATA (buf) != NULL);

  wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
  GST_DEBUG (0, "gst_wavparse_chain: got buffer in '%s'",
          gst_object_get_name (GST_OBJECT (wavparse)));

  size = GST_BUFFER_SIZE (buf);

  /* walk through the states in priority order */
  /* we're in the data region */
  if (wavparse->state == GST_WAVPARSE_DATA) {
    GstFormat format;
    guint64 maxsize;

    /* we can't go beyond the max length */
    maxsize = wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf);

    if (maxsize == 0) {
      return;
    } else if (maxsize < size) {
      /* if we're expected to see a new chunk in this buffer */
      GstBuffer *newbuf;

      newbuf = gst_buffer_create_sub (buf, 0, maxsize);
      gst_buffer_unref (buf);
      buf = newbuf;

      size = maxsize;

      wavparse->state = GST_WAVPARSE_OTHER;
      /* I suppose we could signal an EOF at this point, but that may be
         premature.  We've stopped data flow, that's the main thing. */
    }

    if (GST_PAD_IS_USABLE (wavparse->srcpad)) {
      format = GST_FORMAT_TIME;
      gst_pad_convert (wavparse->srcpad,
		       GST_FORMAT_BYTES,
		       wavparse->offset,
		       &format,
		       &GST_BUFFER_TIMESTAMP (buf));

      if (wavparse->need_discont) {
        if (buf && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
          gst_pad_push (wavparse->srcpad, 
	                GST_BUFFER (gst_event_new_discontinuous (FALSE,
      			      GST_FORMAT_BYTES, wavparse->offset,
			      GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf), 
			      NULL)));
	} else {
          gst_pad_push (wavparse->srcpad, 
	                GST_BUFFER (gst_event_new_discontinuous (FALSE,
      			      GST_FORMAT_BYTES, wavparse->offset,
			      NULL)));
	}
        wavparse->need_discont = FALSE;
      }
      gst_pad_push (wavparse->srcpad, buf);
    } else {
      gst_buffer_unref (buf);
    }

    wavparse->offset += size;

    return;
  }

  if (wavparse->state == GST_WAVPARSE_OTHER) {
    GST_DEBUG (0, "we're in unknown territory here, not passing on");
    return;
  }


  /* here we deal with parsing out the primary state */
  /* these are sequenced such that in the normal case each (RIFF/WAVE,
     fmt, data) will fire in sequence, as they should */

  /* we're in null state now, look for the RIFF header, start parsing */
  if (wavparse->state == GST_WAVPARSE_UNKNOWN) {
    gint retval;

    GST_DEBUG (0, "GstWavParse: checking for RIFF format");

    /* create a new RIFF parser */
    wavparse->riff = gst_riff_new ();

    /* give it the current buffer to start parsing */
    retval = gst_riff_next_buffer (wavparse->riff, buf, 0);
    buffer_riffed = TRUE;
    if (retval < 0) {
      GST_DEBUG (0, "sorry, isn't RIFF");
      return;
    }

    /* this has to be a file of form WAVE for us to deal with it */
    if (wavparse->riff->form != gst_riff_fourcc_to_id ("WAVE")) {
      GST_DEBUG (0, "sorry, isn't WAVE");
      return;
    }

    /* at this point we're waiting for the 'fmt ' chunk */
    wavparse->state = GST_WAVPARSE_CHUNK_FMT;
  }

  /* we're now looking for the 'fmt ' chunk to get the audio info */
  if (wavparse->state == GST_WAVPARSE_CHUNK_FMT) {
    GstRiffChunk *fmt;
    GstWavParseFormat *format;

    GST_DEBUG (0, "GstWavParse: looking for fmt chunk");

    /* there's a good possibility we may not have parsed this buffer */
    if (buffer_riffed == FALSE) {
      gst_riff_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf));
      buffer_riffed = TRUE;
    }

    /* see if the fmt chunk is available yet */
    fmt = gst_riff_get_chunk (wavparse->riff, "fmt ");

    /* if we've got something, deal with it */
    if (fmt != NULL) {
      GstCaps *caps = NULL;

      /* we can gather format information now */
      format = (GstWavParseFormat *)((guchar *) GST_BUFFER_DATA (buf) + fmt->offset);

      wavparse->bps      = GUINT16_FROM_LE(format->wBlockAlign);
      wavparse->rate     = GUINT32_FROM_LE(format->dwSamplesPerSec);
      wavparse->channels = GUINT16_FROM_LE(format->wChannels);
      wavparse->width    = GUINT16_FROM_LE(format->wBitsPerSample);
      wavparse->format	 = GINT16_FROM_LE(format->wFormatTag);

      /* set the caps on the src pad */
      /* FIXME: handle all of the other formats as well */
      switch (wavparse->format)
      {
        case GST_RIFF_WAVE_FORMAT_ALAW:
        case GST_RIFF_WAVE_FORMAT_MULAW:
	  if (!(wavparse->width == 8)) {
	    gst_element_error (GST_ELEMENT (wavparse), "wavparse: invalid wave file");
	    return;
	  }
          caps = GST_CAPS_NEW (
			"parsewav_src",
			"audio/raw",
			"format",	GST_PROPS_STRING ("int"),
			  "law",	GST_PROPS_INT (wavparse->format == GST_RIFF_WAVE_FORMAT_ALAW ? 2 : 1),
			  "endianness",	GST_PROPS_INT (G_LITTLE_ENDIAN),
			  "width",	GST_PROPS_INT (8),
			  "depth",	GST_PROPS_INT (8),
			  "rate",	GST_PROPS_INT (wavparse->rate),
			  "channels",	GST_PROPS_INT (wavparse->channels)
		);
	  break;
        case GST_RIFF_WAVE_FORMAT_PCM:
          caps = GST_CAPS_NEW (
			"parsewav_src",
			"audio/raw",
			"format",	GST_PROPS_STRING ("int"),
			  "law",	GST_PROPS_INT (0),
			  "endianness",	GST_PROPS_INT (G_LITTLE_ENDIAN),
			  "signed",     GST_PROPS_BOOLEAN ((wavparse->width > 8) ? TRUE : FALSE),
			  "width",	GST_PROPS_INT (wavparse->width),
			  "depth",	GST_PROPS_INT (wavparse->width),
			  "rate",	GST_PROPS_INT (wavparse->rate),
			  "channels",	GST_PROPS_INT (wavparse->channels)
		);
	  break;
        case GST_RIFF_WAVE_FORMAT_MPEGL12:
        case GST_RIFF_WAVE_FORMAT_MPEGL3:
	  caps = GST_CAPS_NEW (
			"parsewav_src",
			"audio/x-mp3",
			NULL
		);
	  break;
	default:
          gst_element_error (GST_ELEMENT (wavparse), "wavparse: format %d not handled", wavparse->format);
          return;
      }

      if (gst_pad_try_set_caps (wavparse->srcpad, caps) <= 0) {
        gst_element_error (GST_ELEMENT (wavparse), "Could not set caps");
        return;
      }

      GST_DEBUG (0, "frequency %d, channels %d",
		 wavparse->rate, wavparse->channels);

      /* we're now looking for the data chunk */
      wavparse->state = GST_WAVPARSE_CHUNK_DATA;
    } else {
      /* otherwise we just sort of give up for this buffer */
      gst_buffer_unref (buf);
      return;
    }
  }

  /* now we look for the data chunk */
  if (wavparse->state == GST_WAVPARSE_CHUNK_DATA) {
    GstBuffer *newbuf;
    GstRiffChunk *datachunk;

    GST_DEBUG (0, "GstWavParse: looking for data chunk");

    /* again, we might need to parse the buffer */
    if (buffer_riffed == FALSE) {
      gst_riff_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf));
      buffer_riffed = TRUE;
    }

    datachunk = gst_riff_get_chunk (wavparse->riff, "data");

    if (datachunk != NULL) {
      gulong subsize;

      GST_DEBUG (0, "data begins at %ld", datachunk->offset);

      wavparse->datastart = datachunk->offset;

      /* at this point we can ACK that we have data */
      wavparse->state = GST_WAVPARSE_DATA;

      /* now we construct a new buffer for the remainder */
      subsize = size - datachunk->offset;
      GST_DEBUG (0, "sending last %ld bytes along as audio", subsize);

      newbuf = gst_buffer_create_sub (buf, datachunk->offset, subsize);
      gst_buffer_unref (buf);

      GST_BUFFER_TIMESTAMP (newbuf) = 0;

      if (GST_PAD_IS_USABLE (wavparse->srcpad))
        gst_pad_push (wavparse->srcpad, newbuf);
      else
	gst_buffer_unref (newbuf);

      wavparse->offset = subsize;

      /* now we're ready to go, the next buffer should start data */
      wavparse->state = GST_WAVPARSE_DATA;

      /* however, we may be expecting another chunk at some point */
      wavparse->riff_nextlikely = gst_riff_get_nextlikely (wavparse->riff);
    } else {
      /* otherwise we just sort of give up for this buffer */
      gst_buffer_unref (buf);
      return;
    }
  }
}

/* convert and query stuff */
static const GstFormat *
gst_wavparse_get_formats (GstPad *pad)
{
  static GstFormat formats[] = {
    GST_FORMAT_TIME,
    GST_FORMAT_BYTES,
    GST_FORMAT_UNITS,	/* a "frame", ie a set of samples per Hz */
    0,
    0
  };
  return formats;
}

static gboolean
gst_wavparse_pad_convert (GstPad *pad,
			  GstFormat src_format, gint64 src_value,
			  GstFormat *dest_format, gint64 *dest_value)
{
  gint bytes_per_sample;
  glong byterate;
  GstWavParse *wavparse;

  wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
  /* FIXME default should be samples in this case IMO */
  if (*dest_format == GST_FORMAT_DEFAULT)
    *dest_format = GST_FORMAT_TIME;
  
  bytes_per_sample = wavparse->channels * wavparse->width / 8;
  if (bytes_per_sample == 0) {
    GST_DEBUG (0, "bytes_per_sample is 0, probably an mp3 - channels %d,  width %d\n",
	          wavparse->channels, wavparse->width);
    return FALSE;
  }
  byterate = (glong) (bytes_per_sample * wavparse->rate);
  if (byterate == 0) {
    g_warning ("byterate is 0, internal error\n");
    return FALSE;
  }
  GST_DEBUG (0, "bytes per sample: %d\n", bytes_per_sample);

  switch (src_format) {
    case GST_FORMAT_BYTES:
      if (*dest_format == GST_FORMAT_UNITS)
        *dest_value = src_value / bytes_per_sample;
      else if (*dest_format == GST_FORMAT_TIME)
        *dest_value = src_value * GST_SECOND / byterate;
      else
        return FALSE;
      break;
    case GST_FORMAT_UNITS:
      if (*dest_format == GST_FORMAT_BYTES)
        *dest_value = src_value * bytes_per_sample;
      else if (*dest_format == GST_FORMAT_TIME)
        *dest_value = src_value * GST_SECOND / wavparse->rate;
      else
        return FALSE;
      break;
    case GST_FORMAT_TIME:
      if (*dest_format == GST_FORMAT_BYTES)
	*dest_value = src_value * byterate / GST_SECOND;
      else if (*dest_format == GST_FORMAT_UNITS)
	*dest_value = src_value * wavparse->rate / GST_SECOND;
      else
        return FALSE;

      *dest_value = *dest_value & ~(bytes_per_sample - 1);
      break;
    default:
      g_warning ("unhandled format for wavparse\n");
      break;
  }
  return TRUE;
}
      
static const GstQueryType *
gst_wavparse_get_query_types (GstPad *pad)
{
  static const GstQueryType types[] = {
    GST_QUERY_TOTAL,
    GST_QUERY_POSITION,
    0
  };
  return types;
}

/* handle queries for location and length in requested format */
static gboolean
gst_wavparse_pad_query (GstPad *pad, GstQueryType type,
			GstFormat *format, gint64 *value)
{
  GstFormat peer_format = GST_FORMAT_BYTES;
  gint64 peer_value;
  GstWavParse *wavparse;

  /* probe sink's peer pad, convert value, and that's it :) */
  /* FIXME: ideally we'd loop over possible formats of peer instead
   * of only using BYTE */
  wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
  if (!gst_pad_query (GST_PAD_PEER (wavparse->sinkpad), type, 
		      &peer_format, &peer_value)) {
    g_warning ("Could not query sink pad's peer\n");
    return FALSE;
  }
  if (!gst_pad_convert (wavparse->sinkpad, peer_format, peer_value,
		        format, value)) {
    g_warning ("Could not query sink pad's peer\n");
    return FALSE;
  }
  GST_DEBUG (0, "pad_query done, value %" G_GINT64_FORMAT "\n", *value);
  return TRUE;
}

static const GstEventMask*
gst_wavparse_get_event_masks (GstPad *pad)
{ 
  static const GstEventMask gst_wavparse_src_event_masks[] = {
    { GST_EVENT_SEEK, GST_SEEK_METHOD_SET |
                      GST_SEEK_FLAG_FLUSH },
    { 0, }
  };
  return gst_wavparse_src_event_masks;
}   

static gboolean
gst_wavparse_srcpad_event (GstPad *pad, GstEvent *event)
{
  GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
  gboolean res = FALSE;

  GST_DEBUG(0, "event %d", GST_EVENT_TYPE (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_SEEK:
    {
      gint64 byteoffset;
      GstFormat format;

      /* we can only seek when in the DATA state */
      if (wavparse->state != GST_WAVPARSE_DATA) {
	return FALSE;
      }

      format = GST_FORMAT_BYTES;
      
      /* bring format to bytes for the peer element, 
       * FIXME be smarter here */
      res = gst_pad_convert (pad, 
		             GST_EVENT_SEEK_FORMAT (event),
		             GST_EVENT_SEEK_OFFSET (event),
		             &format,
		             &byteoffset);
      
      if (res) {
	 GstEvent *seek;

	 /* seek to byteoffset + header length */
	 seek = gst_event_new_seek (
			 GST_FORMAT_BYTES |
			 (GST_EVENT_SEEK_TYPE (event) & ~GST_SEEK_FORMAT_MASK), 
			 byteoffset + (GST_EVENT_SEEK_METHOD (event) == GST_SEEK_METHOD_END ? 0 : wavparse->datastart));

	 res = gst_pad_send_event (GST_PAD_PEER (wavparse->sinkpad), seek);

	 if (res) {
           /* ok, seek worked, update our state */
           wavparse->offset = byteoffset; 
	   wavparse->need_discont = TRUE;
	 }
      }
      break;
    }
    default:
      break;
  }

  gst_event_unref (event);
  return res;
}

static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
  GstElementFactory *factory;
  GstTypeFactory *type;

  /* create an elementfactory for the wavparse element */
  factory = gst_element_factory_new ("wavparse", GST_TYPE_WAVPARSE,
                                    &gst_wavparse_details);
  g_return_val_if_fail(factory != NULL, FALSE);
  gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_SECONDARY);

  /* register src pads */
  gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (sink_template_factory));
  gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (src_template_factory));

  gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));

  type = gst_type_factory_new (&wavdefinition);
  gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));

  return TRUE;
}

GstPluginDesc plugin_desc = {
  GST_VERSION_MAJOR,
  GST_VERSION_MINOR,
  "wavparse",
  plugin_init
};