gstreamer/gst/auparse/gstauparse.c

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/* 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.
*/
/* Element-Checklist-Version: 5 */
/* 2001/04/03 - Updated parseau to use caps nego
* Zaheer Abbas Merali <zaheerabbas at merali dot org>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include "gstauparse.h"
#include <gst/audio/audio.h>
/* elementfactory information */
static GstElementDetails gst_au_parse_details =
GST_ELEMENT_DETAILS (".au parser",
"Codec/Demuxer/Audio",
"Parse an .au file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
static GstStaticPadTemplate gst_au_parse_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-au")
);
#define GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS \
"audio/x-alaw, " \
"rate = (int) [ 8000, 192000 ], " \
"channels = (int) [ 1, 2 ]"
#define GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS \
"audio/x-mulaw, " \
"rate = (int) [ 8000, 192000 ], " \
"channels = (int) [ 1, 2 ]"
/* Nothing to decode those ADPCM streams for now */
#define GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS \
"audio/x-adpcm, " \
"layout = (string) { g721, g722, g723_3, g723_5 }"
static GstStaticPadTemplate gst_au_parse_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS ";"
GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS ";"
GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS ";"
GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS));
static void gst_au_parse_dispose (GObject * object);
static GstFlowReturn gst_au_parse_chain (GstPad * pad, GstBuffer * buf);
static GstStateChangeReturn gst_au_parse_change_state (GstElement * element,
GstStateChange transition);
GST_BOILERPLATE (GstAuParse, gst_au_parse, GstElement, GST_TYPE_ELEMENT)
static void gst_au_parse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_au_parse_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_au_parse_src_template));
gst_element_class_set_details (element_class, &gst_au_parse_details);
}
static void
gst_au_parse_class_init (GstAuParseClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_au_parse_dispose;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_au_parse_change_state);
}
static void
gst_au_parse_init (GstAuParse * auparse, GstAuParseClass * klass)
{
auparse->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_au_parse_sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad);
gst_pad_set_chain_function (auparse->sinkpad, gst_au_parse_chain);
auparse->srcpad = gst_pad_new_from_template (gst_static_pad_template_get
(&gst_au_parse_src_template), "src");
gst_pad_use_fixed_caps (auparse->srcpad);
gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad);
auparse->offset = 0;
auparse->buffer_offset = 0;
auparse->adapter = gst_adapter_new ();
auparse->size = 0;
auparse->encoding = 0;
auparse->frequency = 0;
auparse->channels = 0;
}
static void
gst_au_parse_dispose (GObject * object)
{
GstAuParse *au = GST_AU_PARSE (object);
if (au->adapter != NULL) {
g_object_unref (au->adapter);
au->adapter = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstFlowReturn
gst_au_parse_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn ret;
GstAuParse *auparse;
guchar *data;
glong size;
GstCaps *tempcaps;
gint law = 0, depth = 0, ieee = 0;
gchar layout[7];
GstBuffer *subbuf;
GstEvent *event;
layout[0] = 0;
auparse = GST_AU_PARSE (gst_pad_get_parent (pad));
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG_OBJECT (auparse, "got buffer of size %ld", size);
/* if we haven't seen any data yet... */
if (auparse->size == 0) {
guint32 *head = (guint32 *) data;
/* normal format is big endian (au is a Sparc format) */
if (GST_READ_UINT32_BE (head) == 0x2e736e64) { /* ".snd" */
head++;
auparse->le = 0;
auparse->offset = GST_READ_UINT32_BE (head);
head++;
/* Do not trust size, could be set to -1 : unknown */
auparse->size = GST_READ_UINT32_BE (head);
head++;
auparse->encoding = GST_READ_UINT32_BE (head);
head++;
auparse->frequency = GST_READ_UINT32_BE (head);
head++;
auparse->channels = GST_READ_UINT32_BE (head);
head++;
/* and of course, someone had to invent a little endian
* version. Used by DEC systems. */
} else if (GST_READ_UINT32_LE (head) == 0x0064732E) { /* other source say it is "dns." */
head++;
auparse->le = 1;
auparse->offset = GST_READ_UINT32_LE (head);
head++;
/* Do not trust size, could be set to -1 : unknown */
auparse->size = GST_READ_UINT32_LE (head);
head++;
auparse->encoding = GST_READ_UINT32_LE (head);
head++;
auparse->frequency = GST_READ_UINT32_LE (head);
head++;
auparse->channels = GST_READ_UINT32_LE (head);
head++;
} else {
GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, (NULL), (NULL));
gst_buffer_unref (buf);
gst_object_unref (auparse);
return GST_FLOW_ERROR;
}
GST_DEBUG
("offset %ld, size %ld, encoding %ld, frequency %ld, channels %ld\n",
auparse->offset, auparse->size, auparse->encoding, auparse->frequency,
auparse->channels);
/*
Docs :
http://www.opengroup.org/public/pubs/external/auformat.html
http://astronomy.swin.edu.au/~pbourke/dataformats/au/
Solaris headers : /usr/include/audio/au.h
libsndfile : src/au.c
Samples :
http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html
*/
switch (auparse->encoding) {
case 1: /* 8-bit ISDN mu-law G.711 */
law = 1;
depth = 8;
break;
case 27: /* 8-bit ISDN A-law G.711 */
law = 2;
depth = 8;
break;
case 2: /* 8-bit linear PCM */
depth = 8;
break;
case 3: /* 16-bit linear PCM */
depth = 16;
break;
case 4: /* 24-bit linear PCM */
depth = 24;
break;
case 5: /* 32-bit linear PCM */
depth = 32;
break;
case 6: /* 32-bit IEEE floating point */
ieee = 1;
depth = 32;
break;
case 7: /* 64-bit IEEE floating point */
ieee = 1;
depth = 64;
break;
case 23: /* 4-bit CCITT G.721 ADPCM 32kbps -> modplug/libsndfile (compressed 8-bit mu-law) */
strcpy (layout, "g721");
break;
case 24: /* 8-bit CCITT G.722 ADPCM -> rtp */
strcpy (layout, "g722");
break;
case 25: /* 3-bit CCITT G.723.3 ADPCM 24kbps -> rtp/xine/modplug/libsndfile */
strcpy (layout, "g723_3");
break;
case 26: /* 5-bit CCITT G.723.5 ADPCM 40kbps -> rtp/xine/modplug/libsndfile */
strcpy (layout, "g723_5");
break;
case 8: /* Fragmented sample data */
case 9: /* AU_ENCODING_NESTED */
case 10: /* DSP program */
case 11: /* DSP 8-bit fixed point */
case 12: /* DSP 16-bit fixed point */
case 13: /* DSP 24-bit fixed point */
case 14: /* DSP 32-bit fixed point */
case 16: /* AU_ENCODING_DISPLAY : non-audio display data */
case 17: /* AU_ENCODING_MULAW_SQUELCH */
case 18: /* 16-bit linear with emphasis */
case 19: /* 16-bit linear compressed (NeXT) */
case 20: /* 16-bit linear with emphasis and compression */
case 21: /* Music kit DSP commands */
case 22: /* Music kit DSP commands samples */
default:
GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL), (NULL));
gst_buffer_unref (buf);
gst_object_unref (auparse);
return GST_FLOW_ERROR;
}
if (law) {
tempcaps =
gst_caps_new_simple ((law == 1) ? "audio/x-mulaw" : "audio/x-alaw",
"rate", G_TYPE_INT, auparse->frequency,
"channels", G_TYPE_INT, auparse->channels, NULL);
auparse->sample_size = auparse->channels;
} else if (ieee) {
tempcaps = gst_caps_new_simple ("audio/x-raw-float",
"rate", G_TYPE_INT, auparse->frequency,
"channels", G_TYPE_INT, auparse->channels,
"endianness", G_TYPE_INT,
auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN,
"width", G_TYPE_INT, depth, NULL);
auparse->sample_size = auparse->channels * depth / 8;
} else if (layout[0]) {
tempcaps = gst_caps_new_simple ("audio/x-adpcm",
"layout", G_TYPE_STRING, layout, NULL);
auparse->sample_size = 0;
} else {
tempcaps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, auparse->frequency,
"channels", G_TYPE_INT, auparse->channels,
"endianness", G_TYPE_INT,
auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, "depth", G_TYPE_INT,
depth, "width", G_TYPE_INT, depth, "signed", G_TYPE_BOOLEAN, TRUE,
NULL);
auparse->sample_size = auparse->channels * depth / 8;
}
gst_pad_set_active (auparse->srcpad, TRUE);
gst_pad_set_caps (auparse->srcpad, tempcaps);
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_DEFAULT,
0, GST_CLOCK_TIME_NONE, 0);
gst_pad_push_event (auparse->srcpad, event);
subbuf = gst_buffer_create_sub (buf, auparse->offset,
size - auparse->offset);
gst_buffer_unref (buf);
gst_adapter_push (auparse->adapter, subbuf);
} else {
gst_adapter_push (auparse->adapter, buf);
}
if (auparse->sample_size) {
/* Ensure we push a buffer that's a multiple of the frame size downstream */
int avail = gst_adapter_available (auparse->adapter);
avail -= avail % auparse->sample_size;
if (avail > 0) {
const guint8 *data = gst_adapter_peek (auparse->adapter, avail);
GstBuffer *newbuf;
if ((ret =
gst_pad_alloc_buffer_and_set_caps (auparse->srcpad,
auparse->buffer_offset, avail, GST_PAD_CAPS (auparse->srcpad),
&newbuf)) == GST_FLOW_OK) {
memcpy (GST_BUFFER_DATA (newbuf), data, avail);
gst_adapter_flush (auparse->adapter, avail);
auparse->buffer_offset += avail;
ret = gst_pad_push (auparse->srcpad, newbuf);
}
} else
ret = GST_FLOW_OK;
} else {
/* It's something non-trivial (such as ADPCM), we don't understand it, so
* just push downstream and assume this will know what to do with it */
ret = gst_pad_push (auparse->srcpad, buf);
}
gst_object_unref (auparse);
return ret;
}
static GstStateChangeReturn
gst_au_parse_change_state (GstElement * element, GstStateChange transition)
{
GstAuParse *auparse = GST_AU_PARSE (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
ret = parent_class->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
gst_adapter_clear (auparse->adapter);
auparse->buffer_offset = 0;
auparse->offset = 0;
auparse->size = 0;
auparse->encoding = 0;
auparse->frequency = 0;
auparse->channels = 0;
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "auparse", GST_RANK_SECONDARY,
GST_TYPE_AU_PARSE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"auparse",
"parses au streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN)