/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2006> Tim-Philipp Müller <tim centricular net>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-auparse
*
* Parses .au files mostly originating from sun os based computers.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include "gstauparse.h"
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (auparse_debug);
#define GST_CAT_DEFAULT (auparse_debug)
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-au")
);
#define GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS \
"audio/x-raw, " \
"format= (string) { S8, S16LE, S16BE, S24LE, S24BE, " \
"S32LE, S32BE, F32LE, F32BE, " \
"F64LE, F64BE }, " \
"rate = (int) [ 8000, 192000 ], " \
"channels = (int) 1, " \
"layout = (string) interleaved;" \
"audio/x-raw, " \
"format= (string) { S8, S16LE, S16BE, S24LE, S24BE, " \
"S32LE, S32BE, F32LE, F32BE, " \
"F64LE, F64BE }, " \
"rate = (int) [ 8000, 192000 ], " \
"channels = (int) 2, " \
"channel-mask = (bitmask) 0x3," \
"layout = (string) interleaved"
#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 src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AU_PARSE_RAW_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, GstObject * parent,
GstBuffer * buf);
static GstStateChangeReturn gst_au_parse_change_state (GstElement * element,
GstStateChange transition);
static void gst_au_parse_reset (GstAuParse * auparse);
static gboolean gst_au_parse_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_au_parse_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_au_parse_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_au_parse_src_convert (GstAuParse * auparse,
GstFormat src_format, gint64 srcval, GstFormat dest_format,
gint64 * destval);
#define gst_au_parse_parent_class parent_class
G_DEFINE_TYPE (GstAuParse, gst_au_parse, GST_TYPE_ELEMENT);
static void
gst_au_parse_class_init (GstAuParseClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GST_DEBUG_CATEGORY_INIT (auparse_debug, "auparse", 0, ".au parser");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->dispose = gst_au_parse_dispose;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_au_parse_change_state);
gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_set_static_metadata (gstelement_class,
"AU audio demuxer",
"Codec/Demuxer/Audio",
"Parse an .au file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
}
static void
gst_au_parse_init (GstAuParse * auparse)
{
auparse->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (auparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_au_parse_chain));
gst_pad_set_event_function (auparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_au_parse_sink_event));
gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad);
auparse->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_set_query_function (auparse->srcpad,
GST_DEBUG_FUNCPTR (gst_au_parse_src_query));
gst_pad_set_event_function (auparse->srcpad,
GST_DEBUG_FUNCPTR (gst_au_parse_src_event));
gst_pad_use_fixed_caps (auparse->srcpad);
gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad);
auparse->adapter = gst_adapter_new ();
gst_au_parse_reset (auparse);
}
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 void
gst_au_parse_reset (GstAuParse * auparse)
{
auparse->offset = 0;
auparse->buffer_offset = 0;
auparse->encoding = 0;
auparse->samplerate = 0;
auparse->channels = 0;
gst_adapter_clear (auparse->adapter);
gst_caps_replace (&auparse->src_caps, NULL);
/* gst_segment_init (&auparse->segment, GST_FORMAT_TIME); */
}
static void
gst_au_parse_negotiate_srcpad (GstAuParse * auparse, GstCaps * new_caps)
{
if (auparse->src_caps && gst_caps_is_equal (new_caps, auparse->src_caps)) {
GST_LOG_OBJECT (auparse, "same caps, nothing to do");
return;
}
gst_caps_replace (&auparse->src_caps, new_caps);
GST_DEBUG_OBJECT (auparse, "Changing src pad caps to %" GST_PTR_FORMAT,
auparse->src_caps);
gst_pad_set_caps (auparse->srcpad, auparse->src_caps);
return;
}
static GstFlowReturn
gst_au_parse_parse_header (GstAuParse * auparse)
{
GstCaps *tempcaps;
guint32 size;
guint8 *head;
gchar layout[7] = { 0, };
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
gint law = 0;
guint endianness;
head = (guint8 *) gst_adapter_map (auparse->adapter, 24);
g_assert (head != NULL);
GST_DEBUG_OBJECT (auparse, "[%c%c%c%c]", head[0], head[1], head[2], head[3]);
switch (GST_READ_UINT32_BE (head)) {
/* normal format is big endian (au is a Sparc format) */
case 0x2e736e64:{ /* ".snd" */
endianness = G_BIG_ENDIAN;
break;
}
/* and of course, someone had to invent a little endian
* version. Used by DEC systems. */
case 0x646e732e: /* dns. */
case 0x0064732e:{ /* other source say it is "dns." */
endianness = G_LITTLE_ENDIAN;
break;
}
default:{
goto unknown_header;
}
}
auparse->offset = GST_READ_UINT32_BE (head + 4);
/* Do not trust size, could be set to -1 : unknown
* otherwise: filesize = size + auparse->offset
*/
size = GST_READ_UINT32_BE (head + 8);
auparse->encoding = GST_READ_UINT32_BE (head + 12);
auparse->samplerate = GST_READ_UINT32_BE (head + 16);
auparse->channels = GST_READ_UINT32_BE (head + 20);
if (auparse->samplerate < 8000 || auparse->samplerate > 192000)
goto unsupported_sample_rate;
if (auparse->channels < 1 || auparse->channels > 2)
goto unsupported_number_of_channels;
GST_DEBUG_OBJECT (auparse, "offset %" G_GINT64_FORMAT ", size %u, "
"encoding %u, frequency %u, channels %u", auparse->offset, size,
auparse->encoding, auparse->samplerate, 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;
break;
case 27: /* 8-bit ISDN A-law G.711 */
law = 2;
break;
case 2: /* 8-bit linear PCM, FIXME signed? */
format = GST_AUDIO_FORMAT_S8;
auparse->sample_size = auparse->channels;
break;
case 3: /* 16-bit linear PCM */
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_S16LE;
else
format = GST_AUDIO_FORMAT_S16BE;
auparse->sample_size = auparse->channels * 2;
break;
case 4: /* 24-bit linear PCM */
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_S24LE;
else
format = GST_AUDIO_FORMAT_S24BE;
auparse->sample_size = auparse->channels * 3;
break;
case 5: /* 32-bit linear PCM */
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_S32LE;
else
format = GST_AUDIO_FORMAT_S32BE;
auparse->sample_size = auparse->channels * 4;
break;
case 6: /* 32-bit IEEE floating point */
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F32LE;
else
format = GST_AUDIO_FORMAT_F32BE;
auparse->sample_size = auparse->channels * 4;
break;
case 7: /* 64-bit IEEE floating point */
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F64LE;
else
format = GST_AUDIO_FORMAT_F64BE;
auparse->sample_size = auparse->channels * 8;
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:
goto unknown_format;
}
if (law) {
tempcaps =
gst_caps_new_simple ((law == 1) ? "audio/x-mulaw" : "audio/x-alaw",
"rate", G_TYPE_INT, auparse->samplerate,
"channels", G_TYPE_INT, auparse->channels, NULL);
auparse->sample_size = auparse->channels;
} else if (format != GST_AUDIO_FORMAT_UNKNOWN) {
GstCaps *templ_caps = gst_pad_get_pad_template_caps (auparse->srcpad);
GstCaps *intersection;
tempcaps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, gst_audio_format_to_string (format),
"rate", G_TYPE_INT, auparse->samplerate,
"channels", G_TYPE_INT, auparse->channels, NULL);
intersection = gst_caps_intersect (tempcaps, templ_caps);
gst_caps_unref (tempcaps);
gst_caps_unref (templ_caps);
tempcaps = intersection;
} else if (layout[0]) {
tempcaps = gst_caps_new_simple ("audio/x-adpcm",
"layout", G_TYPE_STRING, layout, NULL);
auparse->sample_size = 0;
} else
goto unknown_format;
GST_DEBUG_OBJECT (auparse, "sample_size=%d", auparse->sample_size);
gst_au_parse_negotiate_srcpad (auparse, tempcaps);
GST_DEBUG_OBJECT (auparse, "offset=%" G_GINT64_FORMAT, auparse->offset);
gst_adapter_unmap (auparse->adapter);
gst_adapter_flush (auparse->adapter, auparse->offset);
gst_caps_unref (tempcaps);
return GST_FLOW_OK;
/* ERRORS */
unknown_header:
{
gst_adapter_unmap (auparse->adapter);
GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, (NULL), (NULL));
return GST_FLOW_ERROR;
}
unsupported_sample_rate:
{
gst_adapter_unmap (auparse->adapter);
GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL),
("Unsupported samplerate: %u", auparse->samplerate));
return GST_FLOW_ERROR;
}
unsupported_number_of_channels:
{
gst_adapter_unmap (auparse->adapter);
GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL),
("Unsupported number of channels: %u", auparse->channels));
return GST_FLOW_ERROR;
}
unknown_format:
{
gst_adapter_unmap (auparse->adapter);
GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL),
("Unsupported encoding: %u", auparse->encoding));
return GST_FLOW_ERROR;
}
}
#define AU_HEADER_SIZE 24
static GstFlowReturn
gst_au_parse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstAuParse *auparse;
gint avail, sendnow = 0;
gint64 timestamp;
gint64 duration;
gint64 offset;
auparse = GST_AU_PARSE (parent);
GST_LOG_OBJECT (auparse, "got buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buf));
gst_adapter_push (auparse->adapter, buf);
buf = NULL;
/* if we haven't seen any data yet... */
if (!gst_pad_has_current_caps (auparse->srcpad)) {
if (gst_adapter_available (auparse->adapter) < AU_HEADER_SIZE) {
GST_DEBUG_OBJECT (auparse, "need more data to parse header");
ret = GST_FLOW_OK;
goto out;
}
ret = gst_au_parse_parse_header (auparse);
if (ret != GST_FLOW_OK)
goto out;
if (auparse->need_segment) {
gst_pad_push_event (auparse->srcpad,
gst_event_new_segment (&auparse->segment));
auparse->need_segment = FALSE;
}
}
avail = gst_adapter_available (auparse->adapter);
if (auparse->sample_size > 0) {
/* Ensure we push a buffer that's a multiple of the frame size downstream */
sendnow = avail - (avail % auparse->sample_size);
} else {
/* It's something non-trivial (such as ADPCM), we don't understand it, so
* just push downstream and assume it will know what to do with it */
sendnow = avail;
}
if (sendnow > 0) {
GstBuffer *outbuf;
gint64 pos;
outbuf = gst_adapter_take_buffer (auparse->adapter, sendnow);
outbuf = gst_buffer_make_writable (outbuf);
pos = auparse->buffer_offset - auparse->offset;
pos = MAX (pos, 0);
if (auparse->sample_size > 0 && auparse->samplerate > 0) {
gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos,
GST_FORMAT_DEFAULT, &offset);
gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos,
GST_FORMAT_TIME, ×tamp);
gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES,
sendnow, GST_FORMAT_TIME, &duration);
GST_BUFFER_OFFSET (outbuf) = offset;
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
}
auparse->buffer_offset += sendnow;
ret = gst_pad_push (auparse->srcpad, outbuf);
}
out:
return ret;
}
static gboolean
gst_au_parse_src_convert (GstAuParse * auparse, GstFormat src_format,
gint64 srcval, GstFormat dest_format, gint64 * destval)
{
gboolean ret = TRUE;
guint samplesize, rate;
if (dest_format == src_format) {
*destval = srcval;
return TRUE;
}
GST_OBJECT_LOCK (auparse);
samplesize = auparse->sample_size;
rate = auparse->samplerate;
GST_OBJECT_UNLOCK (auparse);
if (samplesize == 0 || rate == 0) {
GST_LOG_OBJECT (auparse, "cannot convert, sample_size or rate unknown");
return FALSE;
}
switch (src_format) {
case GST_FORMAT_BYTES:
srcval /= samplesize;
/* fallthrough */
case GST_FORMAT_DEFAULT:{
switch (dest_format) {
case GST_FORMAT_DEFAULT:
*destval = srcval;
break;
case GST_FORMAT_BYTES:
*destval = srcval * samplesize;
break;
case GST_FORMAT_TIME:
*destval = gst_util_uint64_scale_int (srcval, GST_SECOND, rate);
break;
default:
ret = FALSE;
break;
}
break;
}
case GST_FORMAT_TIME:{
switch (dest_format) {
case GST_FORMAT_BYTES:
*destval = samplesize *
gst_util_uint64_scale_int (srcval, rate, GST_SECOND);
break;
case GST_FORMAT_DEFAULT:
*destval = gst_util_uint64_scale_int (srcval, rate, GST_SECOND);
break;
default:
ret = FALSE;
break;
}
break;
}
default:{
ret = FALSE;
break;
}
}
if (!ret) {
GST_DEBUG_OBJECT (auparse, "could not convert from %s to %s format",
gst_format_get_name (src_format), gst_format_get_name (dest_format));
}
return ret;
}
static gboolean
gst_au_parse_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstAuParse *auparse;
gboolean ret = FALSE;
auparse = GST_AU_PARSE (parent);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:{
GstFormat format;
gint64 len, val;
gst_query_parse_duration (query, &format, NULL);
if (!gst_pad_peer_query_duration (auparse->sinkpad, GST_FORMAT_BYTES,
&len)) {
GST_DEBUG_OBJECT (auparse, "failed to query upstream length");
break;
}
GST_OBJECT_LOCK (auparse);
len -= auparse->offset;
GST_OBJECT_UNLOCK (auparse);
ret =
gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, len, format,
&val);
if (ret) {
gst_query_set_duration (query, format, val);
}
break;
}
case GST_QUERY_POSITION:{
GstFormat format;
gint64 pos, val;
gst_query_parse_position (query, &format, NULL);
if (!gst_pad_peer_query_position (auparse->sinkpad, GST_FORMAT_BYTES,
&pos)) {
GST_DEBUG_OBJECT (auparse, "failed to query upstream position");
break;
}
GST_OBJECT_LOCK (auparse);
pos -= auparse->offset;
GST_OBJECT_UNLOCK (auparse);
ret = gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, pos,
format, &val);
if (ret) {
gst_query_set_position (query, format, val);
}
break;
}
case GST_QUERY_SEEKING:{
GstFormat format;
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
/* FIXME: query duration in 'format'
gst_query_set_seeking (query, format, TRUE, 0, duration);
*/
gst_query_set_seeking (query, format, TRUE, 0, GST_CLOCK_TIME_NONE);
ret = TRUE;
break;
}
default:
ret = gst_pad_query_default (pad, parent, query);
break;
}
return ret;
}
static gboolean
gst_au_parse_handle_seek (GstAuParse * auparse, GstEvent * event)
{
GstSeekType start_type, stop_type;
GstSeekFlags flags;
GstFormat format;
gdouble rate;
gint64 start, stop;
gboolean res;
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
if (format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (auparse, "only support seeks in TIME format");
return FALSE;
}
res = gst_au_parse_src_convert (auparse, GST_FORMAT_TIME, start,
GST_FORMAT_BYTES, &start);
if (stop > 0) {
res = gst_au_parse_src_convert (auparse, GST_FORMAT_TIME, stop,
GST_FORMAT_BYTES, &stop);
}
GST_INFO_OBJECT (auparse,
"seeking: %" G_GINT64_FORMAT " ... %" G_GINT64_FORMAT, start, stop);
event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, start_type, start,
stop_type, stop);
res = gst_pad_push_event (auparse->sinkpad, event);
return res;
}
static gboolean
gst_au_parse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAuParse *auparse;
gboolean ret = TRUE;
auparse = GST_AU_PARSE (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
/* discard, we'll come up with proper src caps */
gst_event_unref (event);
break;
}
case GST_EVENT_SEGMENT:
{
gint64 start, stop, offset = 0;
GstSegment segment;
/* some debug output */
gst_event_copy_segment (event, &segment);
GST_DEBUG_OBJECT (auparse, "received newsegment %" GST_SEGMENT_FORMAT,
&segment);
start = segment.start;
stop = segment.stop;
if (auparse->sample_size > 0) {
if (start > 0) {
offset = start;
start -= auparse->offset;
start = MAX (start, 0);
}
if (stop > 0) {
stop -= auparse->offset;
stop = MAX (stop, 0);
}
gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, start,
GST_FORMAT_TIME, &start);
gst_au_parse_src_convert (auparse, GST_FORMAT_BYTES, stop,
GST_FORMAT_TIME, &stop);
}
GST_INFO_OBJECT (auparse,
"new segment: %" GST_TIME_FORMAT " ... %" GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.start = segment.time = start;
segment.stop = stop;
gst_segment_copy_into (&segment, &auparse->segment);
if (!gst_pad_has_current_caps (auparse->srcpad)) {
auparse->need_segment = TRUE;
ret = TRUE;
} else {
auparse->need_segment = FALSE;
ret = gst_pad_push_event (auparse->srcpad,
gst_event_new_segment (&segment));
}
auparse->buffer_offset = offset;
gst_event_unref (event);
break;
}
case GST_EVENT_EOS:
if (!auparse->srcpad) {
GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE,
("No valid input found before end of stream"), (NULL));
}
/* fall-through */
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static gboolean
gst_au_parse_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAuParse *auparse;
gboolean ret;
auparse = GST_AU_PARSE (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
ret = gst_au_parse_handle_seek (auparse, event);
gst_event_unref (event);
break;
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
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 = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_au_parse_reset (auparse);
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)