2 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
19 /* Element-Checklist-Version: 5 */
21 /* 2001/04/03 - Updated parseau to use caps nego
22 * Zaheer Abbas Merali <zaheerabbas at merali dot org>
31 #include "gstauparse.h"
32 #include <gst/audio/audio.h>
34 /* elementfactory information */
35 static GstElementDetails gst_auparse_details =
36 GST_ELEMENT_DETAILS (".au parser",
37 "Codec/Demuxer/Audio",
38 "Parse an .au file into raw audio",
39 "Erik Walthinsen <omega@cse.ogi.edu>");
41 static GstStaticPadTemplate gst_auparse_sink_template =
42 GST_STATIC_PAD_TEMPLATE ("sink",
45 GST_STATIC_CAPS ("audio/x-au")
48 static GstStaticPadTemplate gst_auparse_src_template =
49 GST_STATIC_PAD_TEMPLATE ("src",
51 GST_PAD_SOMETIMES, /* FIXME: spider */
52 GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
53 /* we don't use GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS
54 because of min buffer-frames which is 1, not 0 */
56 "rate = (int) [ 1, MAX ], "
57 "channels = (int) [ 1, MAX ], "
58 "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, "
59 "width = (int) { 32, 64 }, "
60 "buffer-frames = (int) [ 0, MAX]" "; "
62 "rate = (int) [ 8000, 192000 ], "
63 "channels = (int) [ 1, 2 ]" "; "
65 "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]" "; "
66 /* Nothing to decode those ADPCM streams for now */
67 "audio/x-adpcm, " "layout = (string) { g721, g722, g723_3, g723_5 }")
76 static void gst_auparse_base_init (gpointer g_class);
77 static void gst_auparse_class_init (GstAuParseClass * klass);
78 static void gst_auparse_init (GstAuParse * auparse);
80 static GstFlowReturn gst_auparse_chain (GstPad * pad, GstBuffer * buf);
82 static GstStateChangeReturn gst_auparse_change_state (GstElement * element,
83 GstStateChange transition);
85 static GstElementClass *parent_class = NULL;
88 /*static guint gst_auparse_signals[LAST_SIGNAL] = { 0 }; */
91 gst_auparse_get_type (void)
93 static GType auparse_type = 0;
96 static const GTypeInfo auparse_info = {
97 sizeof (GstAuParseClass),
98 gst_auparse_base_init,
100 (GClassInitFunc) gst_auparse_class_init,
105 (GInstanceInitFunc) gst_auparse_init,
109 g_type_register_static (GST_TYPE_ELEMENT, "GstAuParse", &auparse_info,
116 gst_auparse_base_init (gpointer g_class)
118 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
120 gst_element_class_add_pad_template (element_class,
121 gst_static_pad_template_get (&gst_auparse_sink_template));
122 gst_element_class_add_pad_template (element_class,
123 gst_static_pad_template_get (&gst_auparse_src_template));
124 gst_element_class_set_details (element_class, &gst_auparse_details);
129 gst_auparse_class_init (GstAuParseClass * klass)
131 GstElementClass *gstelement_class;
133 gstelement_class = (GstElementClass *) klass;
135 parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
137 gstelement_class->change_state = gst_auparse_change_state;
141 gst_auparse_init (GstAuParse * auparse)
144 gst_pad_new_from_template (gst_static_pad_template_get
145 (&gst_auparse_sink_template), "sink");
146 gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad);
147 gst_pad_set_chain_function (auparse->sinkpad, gst_auparse_chain);
149 auparse->srcpad = gst_pad_new_from_template (gst_static_pad_template_get
150 (&gst_auparse_src_template), "src");
151 gst_pad_use_fixed_caps (auparse->srcpad);
152 gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad);
156 auparse->encoding = 0;
157 auparse->frequency = 0;
158 auparse->channels = 0;
162 gst_auparse_chain (GstPad * pad, GstBuffer * buf)
169 gint law = 0, depth = 0, ieee = 0;
174 auparse = GST_AUPARSE (gst_pad_get_parent (pad));
176 GST_DEBUG ("gst_auparse_chain: got buffer in '%s'",
177 gst_element_get_name (GST_ELEMENT (auparse)));
179 data = GST_BUFFER_DATA (buf);
180 size = GST_BUFFER_SIZE (buf);
182 /* if we haven't seen any data yet... */
183 if (auparse->size == 0) {
185 guint32 *head = (guint32 *) data;
187 /* normal format is big endian (au is a Sparc format) */
188 if (GST_READ_UINT32_BE (head) == 0x2e736e64) { /* ".snd" */
191 auparse->offset = GST_READ_UINT32_BE (head);
193 /* Do not trust size, could be set to -1 : unknown */
194 auparse->size = GST_READ_UINT32_BE (head);
196 auparse->encoding = GST_READ_UINT32_BE (head);
198 auparse->frequency = GST_READ_UINT32_BE (head);
200 auparse->channels = GST_READ_UINT32_BE (head);
203 /* and of course, someone had to invent a little endian
204 * version. Used by DEC systems. */
205 } else if (GST_READ_UINT32_LE (head) == 0x0064732E) { /* other source say it is "dns." */
208 auparse->offset = GST_READ_UINT32_LE (head);
210 /* Do not trust size, could be set to -1 : unknown */
211 auparse->size = GST_READ_UINT32_LE (head);
213 auparse->encoding = GST_READ_UINT32_LE (head);
215 auparse->frequency = GST_READ_UINT32_LE (head);
217 auparse->channels = GST_READ_UINT32_LE (head);
221 GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, (NULL), (NULL));
222 gst_buffer_unref (buf);
223 g_object_unref (auparse);
224 return GST_FLOW_ERROR;
228 ("offset %ld, size %ld, encoding %ld, frequency %ld, channels %ld\n",
229 auparse->offset, auparse->size, auparse->encoding, auparse->frequency,
234 http://www.opengroup.org/public/pubs/external/auformat.html
235 http://astronomy.swin.edu.au/~pbourke/dataformats/au/
236 Solaris headers : /usr/include/audio/au.h
237 libsndfile : src/au.c
239 http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html
242 switch (auparse->encoding) {
244 case 1: /* 8-bit ISDN mu-law G.711 */
248 case 27: /* 8-bit ISDN A-law G.711 */
253 case 2: /* 8-bit linear PCM */
256 case 3: /* 16-bit linear PCM */
259 case 4: /* 24-bit linear PCM */
262 case 5: /* 32-bit linear PCM */
266 case 6: /* 32-bit IEEE floating point */
270 case 7: /* 64-bit IEEE floating point */
275 case 23: /* 4-bit CCITT G.721 ADPCM 32kbps -> modplug/libsndfile (compressed 8-bit mu-law) */
276 strcpy (layout, "g721");
278 case 24: /* 8-bit CCITT G.722 ADPCM -> rtp */
279 strcpy (layout, "g722");
281 case 25: /* 3-bit CCITT G.723.3 ADPCM 24kbps -> rtp/xine/modplug/libsndfile */
282 strcpy (layout, "g723_3");
284 case 26: /* 5-bit CCITT G.723.5 ADPCM 40kbps -> rtp/xine/modplug/libsndfile */
285 strcpy (layout, "g723_5");
288 case 8: /* Fragmented sample data */
289 case 9: /* AU_ENCODING_NESTED */
291 case 10: /* DSP program */
292 case 11: /* DSP 8-bit fixed point */
293 case 12: /* DSP 16-bit fixed point */
294 case 13: /* DSP 24-bit fixed point */
295 case 14: /* DSP 32-bit fixed point */
297 case 16: /* AU_ENCODING_DISPLAY : non-audio display data */
298 case 17: /* AU_ENCODING_MULAW_SQUELCH */
300 case 18: /* 16-bit linear with emphasis */
301 case 19: /* 16-bit linear compressed (NeXT) */
302 case 20: /* 16-bit linear with emphasis and compression */
304 case 21: /* Music kit DSP commands */
305 case 22: /* Music kit DSP commands samples */
308 GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL), (NULL));
309 gst_buffer_unref (buf);
310 g_object_unref (auparse);
311 return GST_FLOW_ERROR;
316 gst_caps_new_simple ((law == 1) ? "audio/x-mulaw" : "audio/x-alaw",
317 "rate", G_TYPE_INT, auparse->frequency,
318 "channels", G_TYPE_INT, auparse->channels, NULL);
320 tempcaps = gst_caps_new_simple ("audio/x-raw-float",
321 "rate", G_TYPE_INT, auparse->frequency,
322 "channels", G_TYPE_INT, auparse->channels,
323 "endianness", G_TYPE_INT,
324 auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, "width", G_TYPE_INT,
325 depth, "buffer-frames", G_TYPE_INT, 0, NULL);
326 } else if (layout[0]) {
327 tempcaps = gst_caps_new_simple ("audio/x-adpcm",
328 "layout", G_TYPE_STRING, layout, NULL);
330 tempcaps = gst_caps_new_simple ("audio/x-raw-int",
331 "rate", G_TYPE_INT, auparse->frequency,
332 "channels", G_TYPE_INT, auparse->channels,
333 "endianness", G_TYPE_INT,
334 auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, "depth", G_TYPE_INT,
335 depth, "width", G_TYPE_INT, depth, "signed", G_TYPE_BOOLEAN, TRUE,
339 gst_pad_set_active (auparse->srcpad, TRUE);
340 gst_pad_set_caps (auparse->srcpad, tempcaps);
342 if ((ret = gst_pad_alloc_buffer (auparse->srcpad, GST_BUFFER_OFFSET_NONE,
343 size - (auparse->offset),
344 GST_PAD_CAPS (auparse->srcpad), &newbuf)) != GST_FLOW_OK) {
345 gst_buffer_unref (buf);
346 g_object_unref (auparse);
352 memcpy (GST_BUFFER_DATA (newbuf), data + (auparse->offset),
353 size - (auparse->offset));
354 GST_BUFFER_SIZE (newbuf) = size - (auparse->offset);
360 event = gst_event_new_newsegment (FALSE, 1.0, GST_FORMAT_DEFAULT,
361 0, GST_CLOCK_TIME_NONE, 0);
364 gst_pad_push_event (auparse->srcpad, event);
366 gst_buffer_unref (buf);
367 g_object_unref (auparse);
368 return gst_pad_push (auparse->srcpad, newbuf);
372 g_object_unref (auparse);
373 return gst_pad_push (auparse->srcpad, buf);
377 static GstStateChangeReturn
378 gst_auparse_change_state (GstElement * element, GstStateChange transition)
380 GstAuParse *auparse = GST_AUPARSE (element);
381 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
383 if (parent_class->change_state)
384 ret = parent_class->change_state (element, transition);
386 switch (transition) {
387 case GST_STATE_CHANGE_READY_TO_NULL:
390 auparse->encoding = 0;
391 auparse->frequency = 0;
392 auparse->channels = 0;
401 plugin_init (GstPlugin * plugin)
403 if (!gst_element_register (plugin, "auparse", GST_RANK_SECONDARY,
411 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
414 "parses au streams", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)