gstreamer/gst/wavparse/gstwavparse.c
Tim-Philipp Müller 7448ede81d gst-libs/gst/riff/riff-media.c: Do actually fix invalid RIFF fmt header values for alaw and mulaw audio instead of ju...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Do actually fix invalid RIFF fmt header values for alaw
and mulaw audio instead of just saying so.

* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Give gst_riff_create_audio_caps_with_data() a chance to
fix up broken format header fields before extracting any
parameters from the header. (fixes #167633)
2005-02-20 12:49:19 +00:00

1037 lines
29 KiB
C

/* -*- 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.
*/
#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"
#ifndef G_MAXUINT32
#define G_MAXUINT32 0xffffffff
#endif
GST_DEBUG_CATEGORY (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 GstElementStateReturn gst_wavparse_change_state (GstElement * element);
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_loop (GstElement * element);
static const GstEventMask *gst_wavparse_get_event_masks (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")
);
static GstStaticPadTemplate src_template_factory =
GST_STATIC_PAD_TEMPLATE ("wavparse_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES, /* FIXME: spider */
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) little_endian, "
"signed = (boolean) { true, false }, "
"width = (int) { 8, 16 }, "
"depth = (int) { 8, 16 }, "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 2 ]; "
"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")
);
/* WavParse signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
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),
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_RIFF_READ, "GstWavParse",
&wavparse_info, 0);
}
return wavparse_type;
}
static void
gst_wavparse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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 */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template_factory));
}
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_ref (GST_TYPE_RIFF_READ);
object_class->get_property = gst_wavparse_get_property;
gstelement_class->change_state = gst_wavparse_change_state;
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
}
static void
gst_wavparse_init (GstWavParse * wavparse)
{
/* sink */
wavparse->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&sink_template_factory), "sink");
gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
GST_RIFF_READ (wavparse)->sinkpad = 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);
#if 0
/* source */
wavparse->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&src_template_factory), "src");
gst_pad_use_explicit_caps (wavparse->srcpad);
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);
#endif
gst_element_set_loop_function (GST_ELEMENT (wavparse), gst_wavparse_loop);
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->seek_pending = FALSE;
wavparse->seek_offset = 0;
}
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)
{
gst_wavparse_destroy_sourcepad (wavparse);
/* source */
wavparse->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&src_template_factory), "src");
gst_pad_use_explicit_caps (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);
}
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_stream_init (GstWavParse * wav)
{
GstRiffRead *riff = GST_RIFF_READ (wav);
guint32 doctype;
if (!gst_riff_read_header (riff, &doctype)) {
GST_WARNING_OBJECT (wav, "could not read header");
return FALSE;
}
if (doctype != GST_RIFF_RIFF_WAVE) {
GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL), (NULL));
return FALSE;
}
return TRUE;
}
/* Read 'fmt ' header */
static gboolean
gst_wavparse_fmt (GstWavParse * wav)
{
GstRiffRead *riff = GST_RIFF_READ (wav);
gst_riff_strf_auds *header = NULL;
GstCaps *caps;
if (!gst_riff_read_strf_auds (riff, &header)) {
g_warning ("Not fmt");
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);
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
wav->blockalign = header->blockalign;
wav->width = (header->blockalign * 8) / header->channels;
wav->depth = header->size;
wav->bps = header->av_bps;
g_free (header);
if (caps) {
gst_wavparse_create_sourcepad (wav);
gst_pad_set_explicit_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)
{
GstRiffRead *riff = GST_RIFF_READ (wav);
guint32 tag, length;
if (!gst_riff_peek_head (riff, &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 (riff))) {
GST_WARNING_OBJECT (wav, "could not peek list");
return FALSE;
}
switch (tag) {
case GST_RIFF_LIST_INFO:
if (!gst_riff_read_list (riff, &tag) || !gst_riff_read_info (riff)) {
GST_WARNING_OBJECT (wav, "could not read list");
return FALSE;
}
break;
case GST_RIFF_LIST_adtl:
if (!gst_riff_read_skip (riff)) {
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 (riff)) {
GST_WARNING_OBJECT (wav, "could not read skip");
return FALSE;
}
break;
}
break;
case GST_RIFF_TAG_data:
if (!gst_bytestream_flush (riff->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 (riff->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 (riff->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->dataleft = wav->datasize = (guint64) length;
break;
case GST_RIFF_TAG_cue:
if (!gst_riff_read_skip (riff)) {
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 (riff))
return FALSE;
break;
}
return TRUE;
}
static gboolean
gst_wavparse_handle_seek (GstWavParse * wav)
{
GstRiffRead *riff = GST_RIFF_READ (wav);
GstEvent *event = NULL;
guint32 remaining;
guint8 *data;
if (!gst_bytestream_seek (riff->bs, wav->seek_offset + wav->datastart,
GST_SEEK_METHOD_SET))
return FALSE;
/* wait for discont */
while (!event) {
if (gst_bytestream_peek_bytes (riff->bs, &data, 1)) {
GST_WARNING ("Unexpected data after seek - this means seek failed");
return FALSE;
}
/* get the discont event and return */
gst_bytestream_get_status (riff->bs, &remaining, &event);
if (!event) {
GST_WARNING ("No discontinuity event after seek - seek failed");
return FALSE;
} else if (GST_EVENT_TYPE (event) != GST_EVENT_DISCONTINUOUS) {
GstEventType type = GST_EVENT_TYPE (event);
gst_pad_event_default (riff->sinkpad, event);
if (type == GST_EVENT_EOS)
return FALSE;
event = NULL;
}
}
wav->dataleft = wav->datasize - wav->seek_offset;
gst_event_unref (event);
event = gst_event_new_discontinuous (FALSE,
GST_FORMAT_BYTES, wav->seek_offset,
GST_FORMAT_TIME, GST_SECOND * wav->seek_offset / wav->bps,
GST_FORMAT_UNDEFINED);
gst_pad_event_default (wav->sinkpad, event);
return TRUE;
}
#define MAX_BUFFER_SIZE 4096
static void
gst_wavparse_loop (GstElement * element)
{
GstWavParse *wav = GST_WAVPARSE (element);
GstRiffRead *riff = GST_RIFF_READ (wav);
if (wav->state == GST_WAVPARSE_DATA) {
/* seek handling */
if (wav->seek_pending) {
gst_wavparse_handle_seek (wav);
wav->seek_pending = FALSE;
}
if (wav->dataleft > 0) {
guint32 got_bytes, desired;
GstBuffer *buf = NULL;
desired = MIN (wav->dataleft, MAX_BUFFER_SIZE);
if (!(buf = gst_riff_read_element_data (riff, desired, &got_bytes))) {
GST_WARNING_OBJECT (wav, "trying to read %d bytes failed", desired);
return;
}
GST_DEBUG_OBJECT (wav, "read %d bytes, got %d bytes", desired, got_bytes);
GST_BUFFER_TIMESTAMP (buf) = GST_SECOND *
(wav->datasize - wav->dataleft) / wav->bps;
GST_BUFFER_DURATION (buf) = GST_SECOND * got_bytes / wav->bps;
gst_pad_push (wav->srcpad, GST_DATA (buf));
wav->byteoffset += got_bytes;
if (got_bytes < wav->dataleft) {
wav->dataleft -= got_bytes;
return;
} else {
wav->dataleft = 0;
wav->state = GST_WAVPARSE_OTHER;
}
} else {
wav->state = GST_WAVPARSE_OTHER;
}
}
switch (wav->state) {
case GST_WAVPARSE_START:
if (!gst_wavparse_stream_init (wav)) {
return;
}
wav->state = GST_WAVPARSE_FMT;
/* fall-through */
case GST_WAVPARSE_FMT:
if (!gst_wavparse_fmt (wav)) {
return;
}
wav->state = GST_WAVPARSE_OTHER;
/* fall-through */
case GST_WAVPARSE_OTHER:
if (!gst_wavparse_other (wav)) {
return;
}
break;
case GST_WAVPARSE_DATA:
default:
g_assert_not_reached ();
}
}
/* 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;
}
static gboolean
gst_wavparse_pad_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
guint bytes_per_sample, byterate;
GstWavParse *wavparse;
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
bytes_per_sample = wavparse->channels * wavparse->width / 8;
if (bytes_per_sample == 0) {
GST_DEBUG ("bytes_per_sample 0, probably an mp3 - channels %d, width %d",
wavparse->channels, wavparse->width);
return FALSE;
}
byterate = wavparse->bps;
if (byterate == 0) {
g_warning ("byterate is 0, internal error\n");
return FALSE;
}
GST_DEBUG ("bytes per sample: %d", bytes_per_sample);
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value = src_value * GST_SECOND / byterate;
break;
default:
return FALSE;
}
*dest_value -= *dest_value % bytes_per_sample;
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value = src_value * GST_SECOND / wavparse->rate;
break;
default:
return FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
/* make sure we end up on a sample boundary */
*dest_value =
(src_value * wavparse->rate / GST_SECOND) * wavparse->blockalign;
break;
case GST_FORMAT_DEFAULT:
*dest_value = src_value * wavparse->rate / GST_SECOND;
break;
default:
return FALSE;
}
break;
default:
return FALSE;
}
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)
{
gint64 bytevalue;
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (pad));
/* only if we know */
if (wav->state != GST_WAVPARSE_DATA)
return FALSE;
switch (type) {
case GST_QUERY_POSITION:
bytevalue = wav->datasize - wav->dataleft;
break;
case GST_QUERY_TOTAL:
bytevalue = wav->datasize;
break;
default:
return FALSE;
}
if (*format == GST_FORMAT_BYTES) {
*value = bytevalue;
return TRUE;
}
return gst_pad_convert (wav->sinkpad, GST_FORMAT_BYTES,
bytevalue, format, value);
}
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 ("event %d", GST_EVENT_TYPE (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
gint64 byteoffset;
GstFormat format;
/* bring format to samples for the peer element, */
format = GST_FORMAT_BYTES;
res = gst_pad_convert (pad,
GST_EVENT_SEEK_FORMAT (event),
GST_EVENT_SEEK_OFFSET (event), &format, &byteoffset);
if (res) {
/* ok, seek worked, update our state */
wavparse->seek_offset = byteoffset;
wavparse->seek_pending = TRUE;
}
break;
}
default:
break;
}
gst_event_unref (event);
return res;
}
static GstElementStateReturn
gst_wavparse_change_state (GstElement * element)
{
GstWavParse *wav = GST_WAVPARSE (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
break;
case GST_STATE_READY_TO_PAUSED:
wav->state = GST_WAVPARSE_START;
break;
case GST_STATE_PAUSED_TO_PLAYING:
break;
case GST_STATE_PLAYING_TO_PAUSED:
break;
case GST_STATE_PAUSED_TO_READY:
gst_wavparse_destroy_sourcepad (wav);
wav->state = GST_WAVPARSE_START;
wav->width = 0;
wav->depth = 0;
wav->rate = 0;
wav->channels = 0;
wav->seek_pending = FALSE;
wav->seek_offset = 0;
break;
case GST_STATE_READY_TO_NULL:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_library_load ("riff")) {
return FALSE;
}
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, GST_ORIGIN)