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GST-Tensordecoder-ov_ep/gst/wavparse/gstwavparse.c
Tim-Philipp Müller abb13bdd24 gst/wavparse/gstwavparse.*: Delay source pad creation until we have the first chunk of media data, so the we can exam... 
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
(gst_wavparse_stream_data), (gst_wavparse_loop):
* gst/wavparse/gstwavparse.h:
Delay source pad creation until we have the first chunk of
media data, so the we can examine the data and adjust the
caps accordingly if required. This makes playback of .wav
files with DTS-declared-as-PCM content work (#313266).
2006-03-23 15:36:27 +00:00

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/* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
/* 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.
*/
/**
* SECTION:element-wavparse
*
* <refsect2>
* <para>
* Parse a .wav file into raw or compressed audio.
* </para>
* <para>
* This element currently only supports pull based scheduling.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc sine.wav ! wavparse ! audioconvert ! alsasink
* </programlisting>
* Read a wav file and output to the soundcard using the ALSA element. The
* wav file is assumed to contain raw uncompressed samples.
* </para>
* </refsect2>
*
* Last reviewed on 2006-03-03 (0.10.3)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstwavparse.h"
#include "gst/riff/riff-ids.h"
#include "gst/riff/riff-media.h"
#include <gst/gst-i18n-plugin.h>
GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
#define GST_CAT_DEFAULT (wavparse_debug)
static void gst_wavparse_base_init (gpointer g_class);
static void gst_wavparse_class_init (GstWavParseClass * klass);
static void gst_wavparse_init (GstWavParse * wavparse);
static gboolean gst_wavparse_sink_activate (GstPad * sinkpad);
static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,
gboolean active);
static gboolean gst_wavparse_send_event (GstElement * element,
GstEvent * event);
static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
static gboolean gst_wavparse_pad_convert (GstPad * pad,
GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
static void gst_wavparse_loop (GstPad * pad);
static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
static void gst_wavparse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wav")
);
/* the pad is marked a sometimes and is added to the element when the
* exact type is known. This makes it much easier for a static autoplugger
* to connect the right decoder when needed.
*/
static GstStaticPadTemplate src_template_factory =
GST_STATIC_PAD_TEMPLATE ("wavparse_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) little_endian, "
"signed = (boolean) { true, false }, "
"width = (int) { 8, 16, 24, 32 }, "
"depth = (int) { 8, 16, 24, 32 }, "
"rate = (int) [ 8000, 96000 ], "
"channels = (int) [ 1, 8 ]; "
"audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int) [ 1, 3 ], "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/x-alaw, "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/x-mulaw, "
"rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"
"audio/x-adpcm, "
"layout = (string) microsoft, "
"block_align = (int) [ 1, 8192 ], "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/x-adpcm, "
"layout = (string) dvi, "
"block_align = (int) [ 1, 8192 ], "
"rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"
"audio/x-vnd.sony.atrac3;"
"audio/x-dts;" "audio/x-wma, " "wmaversion = (int) [ 1, 2 ]")
);
static GstElementClass *parent_class = NULL;
GType
gst_wavparse_get_type (void)
{
static GType wavparse_type = 0;
if (!wavparse_type) {
static const GTypeInfo wavparse_info = {
sizeof (GstWavParseClass),
gst_wavparse_base_init,
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_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstPadTemplate *templ;
static GstElementDetails gst_wavparse_details =
GST_ELEMENT_DETAILS (".wav demuxer",
"Codec/Demuxer/Audio",
"Parse a .wav file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
gst_element_class_set_details (element_class, &gst_wavparse_details);
/* register src pads */
templ = gst_static_pad_template_get (&sink_template_factory);
gst_element_class_add_pad_template (element_class, templ);
gst_object_unref (templ);
templ = gst_static_pad_template_get (&src_template_factory);
gst_element_class_add_pad_template (element_class, templ);
gst_object_unref (templ);
}
static void
gst_wavparse_class_init (GstWavParseClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *object_class;
gstelement_class = (GstElementClass *) klass;
object_class = (GObjectClass *) klass;
parent_class = g_type_class_peek_parent (klass);
object_class->get_property = gst_wavparse_get_property;
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
}
static void
gst_wavparse_reset (GstWavParse * wavparse)
{
wavparse->state = GST_WAVPARSE_START;
/* These will all be set correctly in the fmt chunk */
wavparse->depth = 0;
wavparse->rate = 0;
wavparse->width = 0;
wavparse->channels = 0;
wavparse->blockalign = 0;
wavparse->bps = 0;
wavparse->offset = 0;
wavparse->end_offset = 0;
wavparse->dataleft = 0;
wavparse->datasize = 0;
wavparse->datastart = 0;
/* we keep the segment info in time */
gst_segment_init (&wavparse->segment, GST_FORMAT_TIME);
}
static void
gst_wavparse_init (GstWavParse * wavparse)
{
gst_wavparse_reset (wavparse);
/* sink */
wavparse->sinkpad =
gst_pad_new_from_static_template (&sink_template_factory, "sink");
gst_pad_set_activate_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
gst_pad_set_activatepull_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
}
static void
gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
{
if (wavparse->srcpad) {
gst_element_remove_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
wavparse->srcpad = NULL;
}
}
static void
gst_wavparse_create_sourcepad (GstWavParse * wavparse)
{
/* destroy previous one */
gst_wavparse_destroy_sourcepad (wavparse);
/* source */
wavparse->srcpad =
gst_pad_new_from_static_template (&src_template_factory, "src");
gst_pad_use_fixed_caps (wavparse->srcpad);
gst_pad_set_query_type_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_get_query_types));
gst_pad_set_query_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
gst_pad_set_event_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
}
static void
gst_wavparse_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstWavParse *wavparse;
wavparse = GST_WAVPARSE (object);
switch (prop_id) {
default:
break;
}
}
#if 0
static void
gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
{
guint32 got_bytes;
GstByteStream *bs = wavparse->bs;
gst_riff_chunk *temp_chunk, chunk;
guint8 *tempdata;
struct _gst_riff_labl labl, *temp_labl;
struct _gst_riff_ltxt ltxt, *temp_ltxt;
struct _gst_riff_note note, *temp_note;
char *label_name;
GstProps *props;
GstPropsEntry *entry;
GstCaps *new_caps;
GList *caps = NULL;
props = wavparse->metadata->properties;
while (len > 0) {
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
if (got_bytes != sizeof (gst_riff_chunk)) {
return;
}
temp_chunk = (gst_riff_chunk *) tempdata;
chunk.id = GUINT32_FROM_LE (temp_chunk->id);
chunk.size = GUINT32_FROM_LE (temp_chunk->size);
if (chunk.size == 0) {
gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
len -= sizeof (gst_riff_chunk);
continue;
}
switch (chunk.id) {
case GST_RIFF_adtl_labl:
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata,
sizeof (struct _gst_riff_labl));
if (got_bytes != sizeof (struct _gst_riff_labl)) {
return;
}
temp_labl = (struct _gst_riff_labl *) tempdata;
labl.id = GUINT32_FROM_LE (temp_labl->id);
labl.size = GUINT32_FROM_LE (temp_labl->size);
labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
len -= sizeof (struct _gst_riff_labl);
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
if (got_bytes != labl.size - 4) {
return;
}
label_name = (char *) tempdata;
gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
len -= (((labl.size - 4) + 1) & ~1);
new_caps = gst_caps_new ("label",
"application/x-gst-metadata",
gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
"name", G_TYPE_STRING (label_name), NULL));
if (gst_props_get (props, "labels", &caps, NULL)) {
caps = g_list_append (caps, new_caps);
} else {
caps = g_list_append (NULL, new_caps);
entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
gst_props_add_entry (props, entry);
}
break;
case GST_RIFF_adtl_ltxt:
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata,
sizeof (struct _gst_riff_ltxt));
if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
return;
}
temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
len -= sizeof (struct _gst_riff_ltxt);
if (ltxt.size - 20 > 0) {
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
if (got_bytes != ltxt.size - 20) {
return;
}
gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
len -= (((ltxt.size - 20) + 1) & ~1);
label_name = (char *) tempdata;
} else {
label_name = "";
}
new_caps = gst_caps_new ("ltxt",
"application/x-gst-metadata",
gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
"name", G_TYPE_STRING (label_name),
"length", G_TYPE_INT (ltxt.length), NULL));
if (gst_props_get (props, "ltxts", &caps, NULL)) {
caps = g_list_append (caps, new_caps);
} else {
caps = g_list_append (NULL, new_caps);
entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
gst_props_add_entry (props, entry);
}
break;
case GST_RIFF_adtl_note:
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata,
sizeof (struct _gst_riff_note));
if (got_bytes != sizeof (struct _gst_riff_note)) {
return;
}
temp_note = (struct _gst_riff_note *) tempdata;
note.id = GUINT32_FROM_LE (temp_note->id);
note.size = GUINT32_FROM_LE (temp_note->size);
note.identifier = GUINT32_FROM_LE (temp_note->identifier);
gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));
len -= sizeof (struct _gst_riff_note);
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);
if (got_bytes != note.size - 4) {
return;
}
gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);
len -= (((note.size - 4) + 1) & ~1);
label_name = (char *) tempdata;
new_caps = gst_caps_new ("note",
"application/x-gst-metadata",
gst_props_new ("identifier", G_TYPE_INT (note.identifier),
"name", G_TYPE_STRING (label_name), NULL));
if (gst_props_get (props, "notes", &caps, NULL)) {
caps = g_list_append (caps, new_caps);
} else {
caps = g_list_append (NULL, new_caps);
entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));
gst_props_add_entry (props, entry);
}
break;
default:
g_print ("Unknown chunk: %" GST_FOURCC_FORMAT "\n",
GST_FOURCC_ARGS (chunk.id));
return;
}
}
g_object_notify (G_OBJECT (wavparse), "metadata");
}
#endif
#if 0
static void
gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
{
guint32 got_bytes;
GstByteStream *bs = wavparse->bs;
struct _gst_riff_cue *temp_cue, cue;
struct _gst_riff_cuepoints *points;
guint8 *tempdata;
int i;
GList *cues = NULL;
GstPropsEntry *entry;
while (len > 0) {
int required;
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata,
sizeof (struct _gst_riff_cue));
temp_cue = (struct _gst_riff_cue *) tempdata;
/* fixup for our big endian friends */
cue.id = GUINT32_FROM_LE (temp_cue->id);
cue.size = GUINT32_FROM_LE (temp_cue->size);
cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);
gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));
if (got_bytes != sizeof (struct _gst_riff_cue)) {
return;
}
len -= sizeof (struct _gst_riff_cue);
/* -4 because cue.size contains the cuepoints size
and we've already flushed that out of the system */
required = cue.size - 4;
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);
gst_bytestream_flush (bs, ((required) + 1) & ~1);
if (got_bytes != required) {
return;
}
len -= (((cue.size - 4) + 1) & ~1);
/* now we have an array of struct _gst_riff_cuepoints in tempdata */
points = (struct _gst_riff_cuepoints *) tempdata;
for (i = 0; i < cue.cuepoints; i++) {
GstCaps *caps;
caps = gst_caps_new ("cues",
"application/x-gst-metadata",
gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),
"position", G_TYPE_INT (points[i].offset), NULL));
cues = g_list_append (cues, caps);
}
entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));
gst_props_add_entry (wavparse->metadata->properties, entry);
}
g_object_notify (G_OBJECT (wavparse), "metadata");
}
#endif
static gboolean
gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
{
guint32 doctype;
if (!gst_riff_parse_file_header (element, buf, &doctype))
return FALSE;
if (doctype != GST_RIFF_RIFF_WAVE)
goto not_wav;
return TRUE;
/* ERRORS */
not_wav:
{
GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
("File is not an WAVE file: %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (doctype)));
return FALSE;
}
}
static GstFlowReturn
gst_wavparse_stream_init (GstWavParse * wav)
{
GstFlowReturn res;
GstBuffer *buf = NULL;
if ((res = gst_pad_pull_range (wav->sinkpad,
wav->offset, 12, &buf)) != GST_FLOW_OK)
return res;
else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
return GST_FLOW_ERROR;
wav->offset += 12;
return GST_FLOW_OK;
}
#if 0
/* Read 'fmt ' header */
static gboolean
gst_wavparse_fmt (GstWavParse * wav)
{
gst_riff_strf_auds *header = NULL;
GstCaps *caps;
if (!gst_riff_read_strf_auds (wav, &header)) {
g_warning ("Not fmt");
return FALSE;
}
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
if (wav->channels == 0) {
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to contain zero channels - invalid data"));
g_free (header);
return FALSE;
}
wav->blockalign = header->blockalign;
wav->width = (header->blockalign * 8) / header->channels;
wav->depth = header->size;
wav->bps = header->av_bps;
if (wav->bps <= 0) {
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to bitrate of <= zero - invalid data"));
g_free (header);
return FALSE;
}
/* Note: gst_riff_create_audio_caps might nedd to fix values in
* the header header depending on the format, so call it first */
caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);
g_free (header);
if (caps) {
gst_wavparse_create_sourcepad (wav);
gst_pad_use_fixed_caps (wav->srcpad);
gst_pad_set_active (wav->srcpad, TRUE);
gst_pad_set_caps (wav->srcpad, caps);
gst_caps_free (caps);
gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
gst_element_no_more_pads (GST_ELEMENT (wav));
GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
} else {
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
return FALSE;
}
return TRUE;
}
static gboolean
gst_wavparse_other (GstWavParse * wav)
{
guint32 tag, length;
if (!gst_riff_peek_head (wav, &tag, &length, NULL)) {
GST_WARNING_OBJECT (wav, "could not peek head");
return FALSE;
}
GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,
(gchar *) & tag, length);
switch (tag) {
case GST_RIFF_TAG_LIST:
if (!(tag = gst_riff_peek_list (wav))) {
GST_WARNING_OBJECT (wav, "could not peek list");
return FALSE;
}
switch (tag) {
case GST_RIFF_LIST_INFO:
if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) {
GST_WARNING_OBJECT (wav, "could not read list");
return FALSE;
}
break;
case GST_RIFF_LIST_adtl:
if (!gst_riff_read_skip (wav)) {
GST_WARNING_OBJECT (wav, "could not read skip");
return FALSE;
}
break;
default:
GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,
(gchar *) & tag);
if (!gst_riff_read_skip (wav)) {
GST_WARNING_OBJECT (wav, "could not read skip");
return FALSE;
}
break;
}
break;
case GST_RIFF_TAG_data:
if (!gst_bytestream_flush (wav->bs, 8)) {
GST_WARNING_OBJECT (wav, "could not flush 8 bytes");
return FALSE;
}
GST_DEBUG_OBJECT (wav, "switching to data mode");
wav->state = GST_WAVPARSE_DATA;
wav->datastart = gst_bytestream_tell (wav->bs);
if (length == 0) {
guint64 file_length;
/* length is 0, data probably stretches to the end
* of file */
GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");
/* get length of file */
file_length = gst_bytestream_length (wav->bs);
if (file_length == -1) {
GST_DEBUG_OBJECT (wav,
"could not get file length, assuming data to eof");
/* could not get length, assuming till eof */
length = G_MAXUINT32;
}
if (file_length > G_MAXUINT32) {
GST_DEBUG_OBJECT (wav, "file length %lld, clipping to 32 bits");
/* could not get length, assuming till eof */
length = G_MAXUINT32;
} else {
GST_DEBUG_OBJECT (wav, "file length %lld, datalength",
file_length, length);
/* substract offset of datastart from length */
length = file_length - wav->datastart;
GST_DEBUG_OBJECT (wav, "datalength %lld", length);
}
}
wav->datasize = (guint64) length;
break;
case GST_RIFF_TAG_cue:
if (!gst_riff_read_skip (wav)) {
GST_WARNING_OBJECT (wav, "could not read skip");
return FALSE;
}
break;
default:
GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);
if (!gst_riff_read_skip (wav))
return FALSE;
break;
}
return TRUE;
}
#endif
/* This function is used to perform seeks on the element in
* pull mode.
*
* It also works when event is NULL, in which case it will just
* start from the last configured segment. This technique is
* used when activating the element and to perform the seek in
* READY.
*/
static gboolean
gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
{
gboolean res;
gdouble rate;
GstEvent *newsegment;
GstFormat format;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gboolean flush;
gboolean update;
GstSegment seeksegment;
if (event) {
GST_DEBUG_OBJECT (wav, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
/* we have to have a format as the segment format. Try to convert
* if not. */
if (format != GST_FORMAT_TIME) {
GstFormat fmt;
fmt = GST_FORMAT_TIME;
res = TRUE;
if (cur_type != GST_SEEK_TYPE_NONE)
res = gst_pad_query_convert (wav->srcpad, format, cur, &fmt, &cur);
if (res && stop_type != GST_SEEK_TYPE_NONE)
res = gst_pad_query_convert (wav->srcpad, format, stop, &fmt, &stop);
if (!res)
goto no_format;
format = fmt;
}
} else {
GST_DEBUG_OBJECT (wav, "doing seek without event");
flags = 0;
}
flush = flags & GST_SEEK_FLAG_FLUSH;
if (flush)
gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
else
gst_pad_pause_task (wav->sinkpad);
GST_PAD_STREAM_LOCK (wav->sinkpad);
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
if (event) {
GST_DEBUG_OBJECT (wav, "configuring seek");
gst_segment_set_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
if (cur_type != GST_SEEK_TYPE_NONE) {
wav->offset =
gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND);
wav->offset -= wav->offset % wav->bytes_per_sample;
wav->offset += wav->datastart;
}
if (stop != -1) {
wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND);
wav->end_offset +=
wav->bytes_per_sample - (wav->end_offset % wav->bytes_per_sample);
wav->end_offset += wav->datastart;
} else {
wav->end_offset = wav->datasize + wav->datastart;
}
wav->offset = MIN (wav->offset, wav->end_offset);
wav->dataleft = wav->end_offset - wav->offset;
GST_DEBUG_OBJECT (wav,
"seek: offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT ", segment %"
GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, wav->offset, wav->end_offset,
GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
/* prepare for streaming again */
if (flush) {
gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
} else if (wav->segment_running) {
/* we are running the current segment and doing a non-flushing seek,
* close the segment first based on the last_stop. */
GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);
gst_pad_push_event (wav->srcpad,
gst_event_new_new_segment (TRUE,
wav->segment.rate, wav->segment.format,
wav->segment.start, wav->segment.last_stop, wav->segment.time));
}
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT (wav),
gst_message_new_segment_start (GST_OBJECT (wav),
wav->segment.format, wav->segment.last_stop));
}
/* now send the newsegment */
GST_DEBUG_OBJECT (wav, "Sending newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.start, stop);
newsegment =
gst_event_new_new_segment (FALSE, wav->segment.rate,
wav->segment.format, wav->segment.last_stop, stop, wav->segment.time);
if (wav->srcpad) {
gst_pad_push_event (wav->srcpad, newsegment);
} else {
/* send later when we actually create the source pad */
g_assert (wav->newsegment == NULL);
wav->newsegment = newsegment;
}
wav->segment_running = TRUE;
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
wav->sinkpad);
GST_PAD_STREAM_UNLOCK (wav->sinkpad);
return TRUE;
/* ERRORS */
no_format:
{
GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
return FALSE;
}
}
static GstFlowReturn
gst_wavparse_stream_headers (GstWavParse * wav)
{
GstFlowReturn res;
GstBuffer *buf, *extra;
gst_riff_strf_auds *header = NULL;
guint32 tag;
gboolean gotdata = FALSE;
GstCaps *caps;
gint64 duration;
gchar *codec_name = NULL;
GstEvent **event_p;
/* The header start with a 'fmt ' tag */
if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad,
&wav->offset, &tag, &buf)) != GST_FLOW_OK)
return res;
else if (tag != GST_RIFF_TAG_fmt)
goto invalid_wav;
if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra)))
goto parse_header_error;
/* Note: gst_riff_create_audio_caps might nedd to fix values in
* the header header depending on the format, so call it first */
caps =
gst_riff_create_audio_caps (header->format, NULL, header, extra,
NULL, &codec_name);
if (extra)
gst_buffer_unref (extra);
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
if (wav->channels == 0)
goto no_channels;
wav->blockalign = header->blockalign;
wav->width = (header->blockalign * 8) / header->channels;
wav->depth = header->size;
wav->bps = header->av_bps;
if (wav->bps <= 0)
goto no_bitrate;
wav->bytes_per_sample = wav->channels * wav->width / 8;
if (wav->bytes_per_sample <= 0)
goto no_bytes_per_sample;
g_free (header);
if (!caps)
goto unknown_format;
/* create pad later so we can sniff the first few bytes
* of the real data and correct our caps if necessary */
gst_caps_replace (&wav->caps, caps);
gst_caps_replace (&caps, NULL);
if (codec_name) {
wav->tags = gst_tag_list_new ();
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, codec_name, NULL);
g_free (codec_name);
codec_name = NULL;
}
GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate, wav->channels);
/* loop headers until we get data */
while (!gotdata) {
guint size;
guint32 tag;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
&buf)) != GST_FLOW_OK)
goto header_read_error;
/*
wav is a st00pid format, we don't know for sure where data starts.
So we have to go bit by bit until we find the 'data' header
*/
tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
switch (tag) {
/* TODO : Implement the various cases */
case GST_RIFF_TAG_data:
GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
gotdata = TRUE;
wav->offset += 8;
wav->datastart = wav->offset;
wav->datasize = size;
wav->dataleft = size;
wav->end_offset = size + wav->datastart;
break;
default:
GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (tag));
wav->offset += 8 + ((size + 1) & ~1);
}
gst_buffer_unref (buf);
}
GST_DEBUG_OBJECT (wav, "Finished parsing headers");
duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps);
GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, duration);
/* now we have all the info to perform a pending seek if any, if no
* event, this will still do the right thing and it will also send
* the right newsegment event downstream. */
gst_wavparse_perform_seek (wav, wav->seek_event);
/* remove pending event */
event_p = &wav->seek_event;
gst_event_replace (event_p, NULL);
return GST_FLOW_OK;
/* ERROR */
invalid_wav:
{
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
("Invalid WAV header (no fmt at start): %"
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
g_free (codec_name);
return GST_FLOW_ERROR;
}
parse_header_error:
{
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
("Couldn't parse audio header"));
gst_buffer_unref (buf);
g_free (codec_name);
return GST_FLOW_ERROR;
}
no_channels:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to contain no channels - invalid data"));
g_free (header);
g_free (codec_name);
return GST_FLOW_ERROR;
}
no_bitrate:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to have a bitrate of <= zero - invalid data"));
g_free (header);
g_free (codec_name);
return GST_FLOW_ERROR;
}
no_bytes_per_sample:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("could not caluclate bytes per sample - invalid data"));
g_free (header);
g_free (codec_name);
return GST_FLOW_ERROR;
}
unknown_format:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
("No caps found for format 0x%x, %d channels, %d Hz",
wav->format, wav->channels, wav->rate));
g_free (codec_name);
return GST_FLOW_ERROR;
}
header_read_error:
{
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("Couldn't read in header"));
g_free (codec_name);
return GST_FLOW_ERROR;
}
}
/* handle an event sent directly to the element.
*
* This event can be sent either in the READY state or the
* >READY state. The only event of interest really is the seek
* event.
*
* In the READY state we can only store the event and try to
* respect it when going to PAUSED. We assume we are in the
* READY state when our parsing state != GST_WAVPARSE_DATA.
*
* When we are steaming, we can simply perform the seek right
* away.
*/
static gboolean
gst_wavparse_send_event (GstElement * element, GstEvent * event)
{
GstWavParse *wav = GST_WAVPARSE (element);
gboolean res = FALSE;
GstEvent **event_p;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
if (wav->state == GST_WAVPARSE_DATA) {
/* we can handle the seek directly when streaming data */
res = gst_wavparse_perform_seek (wav, event);
} else {
GST_DEBUG_OBJECT (wav, "queuing seek for later");
event_p = &wav->seek_event;
gst_event_replace (event_p, event);
/* we always return true */
res = TRUE;
}
break;
default:
break;
}
gst_event_unref (event);
return res;
}
static void
gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
{
GstStructure *s;
const guint8 dts_marker[] = { 0xFF, 0x1F, 0x00, 0xE8, 0xF1, 0x07 };
s = gst_caps_get_structure (wav->caps, 0);
if (gst_structure_has_name (s, "audio/x-raw-int") &&
GST_BUFFER_SIZE (buf) > 6 &&
memcmp (GST_BUFFER_DATA (buf), dts_marker, 6) == 0) {
GST_WARNING_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
gst_caps_unref (wav->caps);
wav->caps = gst_caps_from_string ("audio/x-dts");
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "dts", NULL);
}
gst_wavparse_create_sourcepad (wav);
gst_pad_set_active (wav->srcpad, TRUE);
gst_pad_set_caps (wav->srcpad, wav->caps);
gst_caps_replace (&wav->caps, NULL);
gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
gst_element_no_more_pads (GST_ELEMENT (wav));
gst_pad_push_event (wav->srcpad, wav->newsegment);
wav->newsegment = NULL;
if (wav->tags) {
gst_element_found_tags_for_pad (GST_ELEMENT (wav), wav->srcpad, wav->tags);
wav->tags = NULL;
}
}
#define MAX_BUFFER_SIZE 4096
static GstFlowReturn
gst_wavparse_stream_data (GstWavParse * wav, gboolean first)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
GstClockTime timestamp, next_timestamp;
guint64 pos, nextpos;
GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT,
wav->offset, wav->end_offset);
/* Get the next n bytes and output them */
if (wav->dataleft == 0)
goto found_eos;
/* scale the amount of data by the segment rate so we get equal
* amounts of data regardless of the playback rate */
desired = MIN (wav->dataleft, MAX_BUFFER_SIZE * ABS (wav->segment.rate));
if (desired >= wav->blockalign && wav->blockalign > 0)
desired -= (desired % wav->blockalign);
GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
"from the sinkpad", desired);
if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
desired, &buf)) != GST_FLOW_OK)
goto pull_error;
obtained = GST_BUFFER_SIZE (buf);
/* first chunk of data? create the source pad. We do this only here so
* we can detect broken .wav files with dts disguised as raw PCM (sigh) */
if (first) {
gst_wavparse_add_src_pad (wav, buf);
}
/* our positions */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
/* update offsets, does not overflow. */
GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
/* and timestamps, be carefull for overflows */
timestamp = gst_util_uint64_scale_int (pos, GST_SECOND, wav->bps);
next_timestamp = gst_util_uint64_scale_int (nextpos, GST_SECOND, wav->bps);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = next_timestamp - timestamp;
/* update current running segment position */
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp);
/* don't forget to set the caps on the buffer */
gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad));
GST_DEBUG_OBJECT (wav,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
", size:%u", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_SIZE (buf));
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
if (obtained < wav->dataleft) {
wav->dataleft -= obtained;
wav->offset += obtained;
} else {
wav->dataleft = 0;
}
return res;
/* ERROR */
found_eos:
{
GST_DEBUG_OBJECT (wav, "found EOS");
/* we completed the segment */
wav->segment_running = FALSE;
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
if ((stop = wav->segment.stop) == -1)
stop = wav->segment.duration;
gst_element_post_message (GST_ELEMENT (wav),
gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME,
stop));
} else {
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
return GST_FLOW_WRONG_STATE;
}
pull_error:
{
GST_DEBUG_OBJECT (wav, "Error getting %ldd bytes from the sinkpad!",
desired);
return res;
}
push_error:
{
GST_DEBUG_OBJECT (wav, "Error pushing on srcpad");
return res;
}
}
static void
gst_wavparse_loop (GstPad * pad)
{
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
switch (wav->state) {
case GST_WAVPARSE_START:
if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
goto pause;
wav->state = GST_WAVPARSE_HEADER;
/* fall-through */
case GST_WAVPARSE_HEADER:
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto pause;
wav->state = GST_WAVPARSE_DATA;
if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK)
goto pause;
break;
case GST_WAVPARSE_DATA:
if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
goto pause;
break;
default:
g_assert_not_reached ();
}
return;
/* ERRORS */
pause:
GST_LOG_OBJECT (wav, "pausing task %d", ret);
gst_pad_pause_task (wav->sinkpad);
if (GST_FLOW_IS_FATAL (ret)) {
/* for fatal errors we post an error message */
GST_ELEMENT_ERROR (wav, STREAM, FAILED,
(_("Internal data stream error.")),
("streaming stopped, reason %s", gst_flow_get_name (ret)));
if (wav->srcpad != NULL)
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
}
#if 0
/* convert and query stuff */
static const GstFormat *
gst_wavparse_get_formats (GstPad * pad)
{
static GstFormat formats[] = {
GST_FORMAT_TIME,
GST_FORMAT_BYTES,
GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
0
};
return formats;
}
#endif
static gboolean
gst_wavparse_pad_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
GstWavParse *wavparse;
gboolean res = TRUE;
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
if (wavparse->bytes_per_sample == 0)
goto no_bytes_per_sample;
if (wavparse->bps == 0)
goto no_bps;
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / wavparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, wavparse->bps);
break;
default:
res = FALSE;
goto done;
}
*dest_value -= *dest_value % wavparse->bytes_per_sample;
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * wavparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, wavparse->rate);
break;
default:
res = FALSE;
goto done;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
/* make sure we end up on a sample boundary */
*dest_value =
gst_util_uint64_scale_int (src_value, wavparse->bps, GST_SECOND);
*dest_value -= *dest_value % wavparse->blockalign;
break;
case GST_FORMAT_DEFAULT:
*dest_value =
gst_util_uint64_scale_int (src_value, wavparse->rate, GST_SECOND);
break;
default:
res = FALSE;
goto done;
}
break;
default:
res = FALSE;
goto done;
}
done:
gst_object_unref (wavparse);
return res;
/* ERRORS */
no_bytes_per_sample:
{
GST_DEBUG_OBJECT (wavparse,
"bytes_per_sample 0, probably an mp3 - channels %d, width %d",
wavparse->channels, wavparse->width);
res = FALSE;
goto done;
}
no_bps:
{
GST_DEBUG_OBJECT (wavparse, "bps 0, cannot convert");
res = FALSE;
goto done;
}
}
static const GstQueryType *
gst_wavparse_get_query_types (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_CONVERT,
0
};
return types;
}
/* handle queries for location and length in requested format */
static gboolean
gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
/* only if we know */
if (wav->state != GST_WAVPARSE_DATA)
return FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
gint64 curb;
gint64 cur;
GstFormat format;
gboolean res = TRUE;
curb = wav->offset - wav->datastart;
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
res &=
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
&format, &cur);
break;
default:
format = GST_FORMAT_BYTES;
cur = curb;
break;
}
if (res)
gst_query_set_position (query, format, cur);
break;
}
case GST_QUERY_DURATION:
{
gint64 endb;
gint64 end;
GstFormat format;
gboolean res = TRUE;
endb = wav->datasize;
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
res &=
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
&format, &end);
break;
default:
format = GST_FORMAT_BYTES;
end = endb;
break;
}
if (res)
gst_query_set_duration (query, format, end);
break;
}
case GST_QUERY_CONVERT:
{
gint64 srcvalue, dstvalue;
GstFormat srcformat, dstformat;
gst_query_parse_convert (query, &srcformat, &srcvalue,
&dstformat, &dstvalue);
res &=
gst_wavparse_pad_convert (pad, srcformat, srcvalue,
&dstformat, &dstvalue);
if (res)
gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
return res;
}
static gboolean
gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
{
GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
gboolean res = TRUE;
GST_DEBUG_OBJECT (wavparse, "event %d", GST_EVENT_TYPE (event));
/* can only handle events when we are in the data state */
if (wavparse->state != GST_WAVPARSE_DATA)
return FALSE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
res = gst_wavparse_perform_seek (wavparse, event);
break;
}
default:
res = FALSE;
break;
}
gst_event_unref (event);
return res;
}
static gboolean
gst_wavparse_sink_activate (GstPad * sinkpad)
{
if (gst_pad_check_pull_range (sinkpad))
return gst_pad_activate_pull (sinkpad, TRUE);
/* FIXME, we can only operate in pull mode for now */
GST_DEBUG_OBJECT (sinkpad, "pull_range not supported on sinkpad");
return FALSE;
};
static gboolean
gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
if (active) {
gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop, sinkpad);
} else {
gst_pad_stop_task (sinkpad);
}
gst_object_unref (wav);
return TRUE;
};
static GstStateChangeReturn
gst_wavparse_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstWavParse *wav = GST_WAVPARSE (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_wavparse_reset (wav);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:{
GstEvent **event_p = &wav->seek_event;
gst_wavparse_destroy_sourcepad (wav);
gst_event_replace (event_p, NULL);
gst_wavparse_reset (wav);
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
gst_riff_init ();
return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
GST_TYPE_WAVPARSE);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"wavparse",
"Parse a .wav file into raw audio",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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