mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-08 18:39:54 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/wavparse/gstwavparse.c
Benjamin Otte 98cfd56ea3 bugfixes: 
Original commit message from CVS:
bugfixes:
- seek correctly on SEEK_METHOD_END
- don't emit a warning when mp3's in wav's have no width information
- use BYTES format on discontinuous events and omit timestamp when invalid (mp3 anyone?)
2003-04-29 21:54:47 +00:00

689 lines
20 KiB
C
Raw Blame History

/* 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.
*/
#include <string.h>
#include <gstwavparse.h>
static void gst_wavparse_class_init (GstWavParseClass *klass);
static void gst_wavparse_init (GstWavParse *wavparse);
static GstCaps* wav_type_find (GstBuffer *buf, gpointer private);
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_chain (GstPad *pad, GstBuffer *buf);
static const GstEventMask*
gst_wavparse_get_event_masks (GstPad *pad);
static gboolean gst_wavparse_srcpad_event (GstPad *pad, GstEvent *event);
/* elementfactory information */
static GstElementDetails gst_wavparse_details = {
".wav parser",
"Codec/Parser",
"LGPL",
"Parse a .wav file into raw audio",
VERSION,
"Erik Walthinsen <omega@cse.ogi.edu>",
"(C) 1999",
};
GST_PAD_TEMPLATE_FACTORY (sink_template_factory,
"wavparse_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"wavparse_wav",
"audio/x-wav",
NULL
)
)
GST_PAD_TEMPLATE_FACTORY (src_template_factory,
"wavparse_src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"wavparse_raw",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_INT (G_LITTLE_ENDIAN),
"signed", GST_PROPS_LIST (
GST_PROPS_BOOLEAN (FALSE),
GST_PROPS_BOOLEAN (TRUE)
),
"width", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16)
),
"depth", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16)
),
"rate", GST_PROPS_INT_RANGE (8000, 48000),
"channels", GST_PROPS_INT_RANGE (1, 2)
),
GST_CAPS_NEW (
"wavparse_mp3",
"audio/x-mp3",
NULL
),
GST_CAPS_NEW (
"parsewav_law",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT_RANGE (1, 2),
"endianness", GST_PROPS_INT (G_LITTLE_ENDIAN),
"width", GST_PROPS_INT (8),
"depth", GST_PROPS_INT (8),
"rate", GST_PROPS_INT_RANGE (8000, 48000),
"channels", GST_PROPS_INT_RANGE (1, 2)
)
)
/* typefactory for 'wav' */
static GstTypeDefinition
wavdefinition =
{
"wavparse_audio/x-wav",
"audio/x-wav",
".wav",
wav_type_find,
};
/* WavParse signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
/* FILL ME */
};
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), NULL,
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_class_init (GstWavParseClass *klass)
{
GstElementClass *gstelement_class;
gstelement_class = (GstElementClass*) klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
}
static void
gst_wavparse_init (GstWavParse *wavparse)
{
/* sink */
wavparse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_template_factory), "sink");
gst_element_add_pad (GST_ELEMENT (wavparse), 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);
/* source */
wavparse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_template_factory), "src");
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);
gst_pad_set_chain_function (wavparse->sinkpad, gst_wavparse_chain);
wavparse->riff = NULL;
wavparse->state = GST_WAVPARSE_UNKNOWN;
wavparse->riff = NULL;
wavparse->riff_nextlikely = 0;
wavparse->size = 0;
wavparse->bps = 0;
wavparse->offset = 0;
wavparse->need_discont = FALSE;
}
static GstCaps*
wav_type_find (GstBuffer *buf, gpointer private)
{
gchar *data = GST_BUFFER_DATA (buf);
if (GST_BUFFER_SIZE (buf) < 12) return NULL;
if (strncmp (&data[0], "RIFF", 4)) return NULL;
if (strncmp (&data[8], "WAVE", 4)) return NULL;
return gst_caps_new ("wav_type_find", "audio/x-wav", NULL);
}
static void
gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
{
GstWavParse *wavparse;
gboolean buffer_riffed = FALSE; /* so we don't parse twice */
gulong size;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
g_return_if_fail (GST_BUFFER_DATA (buf) != NULL);
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
GST_DEBUG (0, "gst_wavparse_chain: got buffer in '%s'",
gst_object_get_name (GST_OBJECT (wavparse)));
size = GST_BUFFER_SIZE (buf);
/* walk through the states in priority order */
/* we're in the data region */
if (wavparse->state == GST_WAVPARSE_DATA) {
GstFormat format;
guint64 maxsize;
/* we can't go beyond the max length */
maxsize = wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf);
/* if we're expected to see a new chunk in this buffer */
if (maxsize < size) {
GstBuffer *newbuf;
newbuf = gst_buffer_create_sub (buf, 0, maxsize);
gst_buffer_unref (buf);
buf = newbuf;
size = maxsize;
wavparse->state = GST_WAVPARSE_OTHER;
/* I suppose we could signal an EOF at this point, but that may be
premature. We've stopped data flow, that's the main thing. */
}
if (GST_PAD_IS_USABLE (wavparse->srcpad)) {
format = GST_FORMAT_TIME;
gst_pad_convert (wavparse->srcpad,
GST_FORMAT_BYTES,
wavparse->offset,
&format,
&GST_BUFFER_TIMESTAMP (buf));
if (wavparse->need_discont) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
gst_pad_push (wavparse->srcpad,
GST_BUFFER (gst_event_new_discontinuous (FALSE,
GST_FORMAT_BYTES, wavparse->offset,
GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf),
NULL)));
} else {
gst_pad_push (wavparse->srcpad,
GST_BUFFER (gst_event_new_discontinuous (FALSE,
GST_FORMAT_BYTES, wavparse->offset,
NULL)));
}
wavparse->need_discont = FALSE;
}
gst_pad_push (wavparse->srcpad, buf);
}
else
gst_buffer_unref (buf);
wavparse->offset += size;
return;
}
if (wavparse->state == GST_WAVPARSE_OTHER) {
GST_DEBUG (0, "we're in unknown territory here, not passing on");
return;
}
/* here we deal with parsing out the primary state */
/* these are sequenced such that in the normal case each (RIFF/WAVE,
fmt, data) will fire in sequence, as they should */
/* we're in null state now, look for the RIFF header, start parsing */
if (wavparse->state == GST_WAVPARSE_UNKNOWN) {
gint retval;
GST_DEBUG (0, "GstWavParse: checking for RIFF format");
/* create a new RIFF parser */
wavparse->riff = gst_riff_new ();
/* give it the current buffer to start parsing */
retval = gst_riff_next_buffer (wavparse->riff, buf, 0);
buffer_riffed = TRUE;
if (retval < 0) {
GST_DEBUG (0, "sorry, isn't RIFF");
return;
}
/* this has to be a file of form WAVE for us to deal with it */
if (wavparse->riff->form != gst_riff_fourcc_to_id ("WAVE")) {
GST_DEBUG (0, "sorry, isn't WAVE");
return;
}
/* at this point we're waiting for the 'fmt ' chunk */
wavparse->state = GST_WAVPARSE_CHUNK_FMT;
}
/* we're now looking for the 'fmt ' chunk to get the audio info */
if (wavparse->state == GST_WAVPARSE_CHUNK_FMT) {
GstRiffChunk *fmt;
GstWavParseFormat *format;
GST_DEBUG (0, "GstWavParse: looking for fmt chunk");
/* there's a good possibility we may not have parsed this buffer */
if (buffer_riffed == FALSE) {
gst_riff_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf));
buffer_riffed = TRUE;
}
/* see if the fmt chunk is available yet */
fmt = gst_riff_get_chunk (wavparse->riff, "fmt ");
/* if we've got something, deal with it */
if (fmt != NULL) {
GstCaps *caps = NULL;
/* we can gather format information now */
format = (GstWavParseFormat *)((guchar *) GST_BUFFER_DATA (buf) + fmt->offset);
wavparse->bps = GUINT16_FROM_LE(format->wBlockAlign);
wavparse->rate = GUINT32_FROM_LE(format->dwSamplesPerSec);
wavparse->channels = GUINT16_FROM_LE(format->wChannels);
wavparse->width = GUINT16_FROM_LE(format->wBitsPerSample);
wavparse->format = GINT16_FROM_LE(format->wFormatTag);
/* set the caps on the src pad */
/* FIXME: handle all of the other formats as well */
switch (wavparse->format)
{
case GST_RIFF_WAVE_FORMAT_ALAW:
case GST_RIFF_WAVE_FORMAT_MULAW:
if (!(wavparse->width == 8)) {
gst_element_error (GST_ELEMENT (wavparse), "wavparse: invalid wave file");
return;
}
caps = GST_CAPS_NEW (
"parsewav_src",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (wavparse->format == GST_RIFF_WAVE_FORMAT_ALAW ? 2 : 1),
"endianness", GST_PROPS_INT (G_LITTLE_ENDIAN),
"width", GST_PROPS_INT (8),
"depth", GST_PROPS_INT (8),
"rate", GST_PROPS_INT (wavparse->rate),
"channels", GST_PROPS_INT (wavparse->channels)
);
break;
case GST_RIFF_WAVE_FORMAT_PCM:
caps = GST_CAPS_NEW (
"parsewav_src",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_INT (G_LITTLE_ENDIAN),
"signed", GST_PROPS_BOOLEAN ((wavparse->width > 8) ? TRUE : FALSE),
"width", GST_PROPS_INT (wavparse->width),
"depth", GST_PROPS_INT (wavparse->width),
"rate", GST_PROPS_INT (wavparse->rate),
"channels", GST_PROPS_INT (wavparse->channels)
);
break;
case GST_RIFF_WAVE_FORMAT_MPEGL12:
case GST_RIFF_WAVE_FORMAT_MPEGL3:
caps = GST_CAPS_NEW (
"parsewav_src",
"audio/x-mp3",
NULL
);
break;
default:
gst_element_error (GST_ELEMENT (wavparse), "wavparse: format %d not handled", wavparse->format);
return;
}
if (gst_pad_try_set_caps (wavparse->srcpad, caps) <= 0) {
gst_element_error (GST_ELEMENT (wavparse), "Could not set caps");
return;
}
GST_DEBUG (0, "frequency %d, channels %d",
wavparse->rate, wavparse->channels);
/* we're now looking for the data chunk */
wavparse->state = GST_WAVPARSE_CHUNK_DATA;
} else {
/* otherwise we just sort of give up for this buffer */
gst_buffer_unref (buf);
return;
}
}
/* now we look for the data chunk */
if (wavparse->state == GST_WAVPARSE_CHUNK_DATA) {
GstBuffer *newbuf;
GstRiffChunk *datachunk;
GST_DEBUG (0, "GstWavParse: looking for data chunk");
/* again, we might need to parse the buffer */
if (buffer_riffed == FALSE) {
gst_riff_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf));
buffer_riffed = TRUE;
}
datachunk = gst_riff_get_chunk (wavparse->riff, "data");
if (datachunk != NULL) {
gulong subsize;
GST_DEBUG (0, "data begins at %ld", datachunk->offset);
wavparse->datastart = datachunk->offset;
/* at this point we can ACK that we have data */
wavparse->state = GST_WAVPARSE_DATA;
/* now we construct a new buffer for the remainder */
subsize = size - datachunk->offset;
GST_DEBUG (0, "sending last %ld bytes along as audio", subsize);
newbuf = gst_buffer_create_sub (buf, datachunk->offset, subsize);
gst_buffer_unref (buf);
GST_BUFFER_TIMESTAMP (newbuf) = 0;
if (GST_PAD_IS_USABLE (wavparse->srcpad))
gst_pad_push (wavparse->srcpad, newbuf);
else
gst_buffer_unref (newbuf);
wavparse->offset = subsize;
/* now we're ready to go, the next buffer should start data */
wavparse->state = GST_WAVPARSE_DATA;
/* however, we may be expecting another chunk at some point */
wavparse->riff_nextlikely = gst_riff_get_nextlikely (wavparse->riff);
} else {
/* otherwise we just sort of give up for this buffer */
gst_buffer_unref (buf);
return;
}
}
}
/* convert and query stuff */
static const GstFormat *
gst_wavparse_get_formats (GstPad *pad)
{
static GstFormat formats[] = {
GST_FORMAT_TIME,
GST_FORMAT_BYTES,
GST_FORMAT_UNITS, /* a "frame", ie a set of samples per Hz */
0,
0
};
return formats;
}
static gboolean
gst_wavparse_pad_convert (GstPad *pad,
GstFormat src_format, gint64 src_value,
GstFormat *dest_format, gint64 *dest_value)
{
gint bytes_per_sample;
glong byterate;
GstWavParse *wavparse;
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
/* FIXME default should be samples in this case IMO */
if (*dest_format == GST_FORMAT_DEFAULT)
*dest_format = GST_FORMAT_TIME;
bytes_per_sample = wavparse->channels * wavparse->width / 8;
if (bytes_per_sample == 0) {
GST_DEBUG (0, "bytes_per_sample is 0, probably an mp3 - channels %d, width %d\n",
wavparse->channels, wavparse->width);
return FALSE;
}
byterate = (glong) (bytes_per_sample * wavparse->rate);
if (byterate == 0) {
g_warning ("byterate is 0, internal error\n");
return FALSE;
}
GST_DEBUG (0, "bytes per sample: %d\n", bytes_per_sample);
switch (src_format) {
case GST_FORMAT_BYTES:
if (*dest_format == GST_FORMAT_UNITS)
*dest_value = src_value / bytes_per_sample;
else if (*dest_format == GST_FORMAT_TIME)
*dest_value = src_value * GST_SECOND / byterate;
else
return FALSE;
break;
case GST_FORMAT_UNITS:
if (*dest_format == GST_FORMAT_BYTES)
*dest_value = src_value * bytes_per_sample;
else if (*dest_format == GST_FORMAT_TIME)
*dest_value = src_value * GST_SECOND / wavparse->rate;
else
return FALSE;
break;
case GST_FORMAT_TIME:
if (*dest_format == GST_FORMAT_BYTES)
*dest_value = src_value * byterate / GST_SECOND;
else if (*dest_format == GST_FORMAT_UNITS)
*dest_value = src_value * wavparse->rate / GST_SECOND;
else
return FALSE;
*dest_value = *dest_value & ~(bytes_per_sample - 1);
break;
default:
g_warning ("unhandled format for wavparse\n");
break;
}
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)
{
GstFormat peer_format = GST_FORMAT_BYTES;
gint64 peer_value;
GstWavParse *wavparse;
/* probe sink's peer pad, convert value, and that's it :) */
/* FIXME: ideally we'd loop over possible formats of peer instead
* of only using BYTE */
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
if (!gst_pad_query (GST_PAD_PEER (wavparse->sinkpad), type,
&peer_format, &peer_value)) {
g_warning ("Could not query sink pad's peer\n");
return FALSE;
}
if (!gst_pad_convert (wavparse->sinkpad, peer_format, peer_value,
format, value)) {
g_warning ("Could not query sink pad's peer\n");
return FALSE;
}
GST_DEBUG (0, "pad_query done, value %" G_GINT64_FORMAT "\n", *value);
return TRUE;
}
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(0, "event %d", GST_EVENT_TYPE (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
gint64 byteoffset;
GstFormat format;
/* we can only seek when in the DATA state */
if (wavparse->state != GST_WAVPARSE_DATA) {
return FALSE;
}
format = GST_FORMAT_BYTES;
/* bring format to bytes for the peer element,
* FIXME be smarter here */
res = gst_pad_convert (pad,
GST_EVENT_SEEK_FORMAT (event),
GST_EVENT_SEEK_OFFSET (event),
&format,
&byteoffset);
if (res) {
GstEvent *seek;
/* seek to byteoffset + header length */
seek = gst_event_new_seek (
GST_FORMAT_BYTES |
(GST_EVENT_SEEK_TYPE (event) & ~GST_SEEK_FORMAT_MASK),
byteoffset + (GST_EVENT_SEEK_METHOD (event) == GST_SEEK_METHOD_END ? 0 : wavparse->datastart));
res = gst_pad_send_event (GST_PAD_PEER (wavparse->sinkpad), seek);
if (res) {
/* ok, seek worked, update our state */
wavparse->offset = byteoffset;
wavparse->need_discont = TRUE;
}
}
break;
}
default:
break;
}
gst_event_unref (event);
return res;
}
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
GstTypeFactory *type;
/* create an elementfactory for the wavparse element */
factory = gst_element_factory_new ("wavparse", GST_TYPE_WAVPARSE,
&gst_wavparse_details);
g_return_val_if_fail(factory != NULL, FALSE);
gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_SECONDARY);
/* register src pads */
gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (sink_template_factory));
gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (src_template_factory));
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
type = gst_type_factory_new (&wavdefinition);
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"wavparse",
plugin_init
};
Reference in a new issue View git blame Copy permalink