mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-26 02:00:33 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/wavparse/gstwavparse.c
Tim-Philipp Müller 3cbd39e022 Update for stream lock API changes: don't take stream log in sink event handlers any longer and change GST_STREAM_LOC... 
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event):
* ext/flac/gstflacdec.c: (gst_flacdec_loop),
(gst_flacdec_src_event):
* ext/flac/gstflacenc.c: (gst_flacenc_sink_event):
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_event),
(gst_signal_processor_getrange), (gst_signal_processor_chain):
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
* gst/flx/gstflxdec.c: (gst_flxdec_src_event_handler),
(gst_flxdec_sink_event_handler):
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_handle_seek_event):
* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek):
Update for stream lock API changes: don't take stream log
in sink event handlers any longer and change GST_STREAM_LOCK
to GST_PAD_STREAM_LOCK. Don't leak references in flxdec event
functions.
2005-11-21 18:09:02 +00:00

1397 lines
38 KiB
C
Raw Blame History

/* -*- 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"
#include <gst/gst-i18n-plugin.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 gboolean gst_wavparse_sink_activate (GstPad * sinkpad);
static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,
gboolean active);
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")
);
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, 24, 32 }, "
"depth = (int) { 8, 16, 24, 32 }, "
"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_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);
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_ELEMENT);
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_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->dataleft = 0;
wavparse->datasize = 0;
wavparse->datastart = 0;
if (wavparse->seek_event)
gst_event_unref (wavparse->seek_event);
wavparse->seek_event = NULL;
wavparse->seek_pending = FALSE;
wavparse->segment_rate = 1.0;
wavparse->segment_start = -1;
wavparse->segment_stop = -1;
}
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_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_wavparse_reset (wavparse);
}
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_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;
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->dataleft = 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
static gboolean
gst_wavparse_handle_seek (GstWavParse * wav, gboolean update)
{
GstClockTime start_time, stop_time;
gboolean flush;
flush = wav->segment_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);
if (update) {
wav->offset = wav->segment_start + wav->datastart;
wav->dataleft = wav->segment_stop - wav->segment_start;
start_time = GST_SECOND * wav->segment_start / wav->bps;
} else {
start_time = (wav->offset - wav->datastart) * GST_SECOND / wav->bps;
}
stop_time = GST_SECOND * wav->segment_stop / wav->bps;
GST_DEBUG ("seek: offset %" G_GUINT64_FORMAT ", len %" G_GUINT64_FORMAT
", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT,
wav->offset, wav->dataleft, GST_TIME_ARGS (start_time),
GST_TIME_ARGS (stop_time));
wav->seek_event = gst_event_new_newsegment (!update, wav->segment_rate,
GST_FORMAT_TIME, start_time, stop_time, start_time);
if (flush)
gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
if (wav->segment_flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT (wav),
gst_message_new_segment_start (GST_OBJECT (wav), GST_FORMAT_TIME,
start_time));
}
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
wav->sinkpad);
GST_PAD_STREAM_UNLOCK (wav->sinkpad);
return TRUE;
}
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;
/* 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, NULL,
NULL, NULL);
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;
gst_wavparse_create_sourcepad (wav);
gst_pad_set_active (wav->srcpad, TRUE);
gst_pad_set_caps (wav->srcpad, caps);
gst_caps_unref (caps);
caps = NULL;
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);
/* 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 ("Got 'data' TAG, size : %d", size);
gotdata = TRUE;
wav->offset += 8;
wav->datastart = wav->offset;
wav->datasize = size;
wav->dataleft = wav->datasize;
break;
default:
GST_DEBUG ("Ignoring tag %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag));
wav->offset += 8 + ((size + 1) & ~1);
}
gst_buffer_unref (buf);
}
GST_DEBUG ("Finished parsing headers");
wav->segment_start = 0;
/* FIXME, can overflow */
wav->segment_stop = (gint64) GST_SECOND *wav->datasize / wav->bps;
/* Initial discont */
wav->seek_event = gst_event_new_newsegment (FALSE, 1.0,
GST_FORMAT_TIME,
wav->segment_start, wav->segment_stop, wav->segment_start);
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)));
return GST_FLOW_ERROR;
}
parse_header_error:
{
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
("Couldn't parse audio header"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
no_channels:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to contain no channels - invalid data"));
g_free (header);
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);
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);
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));
return GST_FLOW_ERROR;
}
header_read_error:
{
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("Couldn't read in header"));
return GST_FLOW_ERROR;
}
}
#define MAX_BUFFER_SIZE 4096
static GstFlowReturn
gst_wavparse_stream_data (GstWavParse * wav)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
GST_DEBUG ("offset : %lld , dataleft : %lld", wav->offset, wav->dataleft);
/* Get the next n bytes and output them */
if (wav->dataleft == 0)
goto found_eos;
desired = MIN (wav->dataleft, MAX_BUFFER_SIZE * ABS (wav->segment_rate));
if (desired >= wav->blockalign && wav->blockalign > 0)
desired -= (desired % wav->blockalign);
GST_DEBUG ("Fetching %lld 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);
GST_BUFFER_OFFSET (buf) =
(wav->offset - wav->datastart) / wav->bytes_per_sample;
GST_BUFFER_TIMESTAMP (buf) =
GST_SECOND * (wav->offset - wav->datastart) / wav->bps;
GST_BUFFER_DURATION (buf) = 1 + GST_SECOND * obtained / wav->bps;
gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad));
GST_DEBUG ("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 ("found EOS");
if (wav->segment_flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop_time;
stop_time = GST_SECOND * wav->segment_stop / wav->bps;
gst_element_post_message (GST_ELEMENT (wav),
gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME,
stop_time));
} else {
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
return GST_FLOW_WRONG_STATE;
}
pull_error:
{
GST_DEBUG ("Error getting %ldd bytes from the sinkpad!", desired);
return res;
}
push_error:
{
GST_DEBUG ("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;
/* fall-through */
case GST_WAVPARSE_DATA:
if (wav->seek_event) {
gst_pad_push_event (wav->srcpad, wav->seek_event);
wav->seek_event = NULL;
}
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto pause;
break;
default:
g_assert_not_reached ();
}
return;
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)));
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 = 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 = 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 =
(src_value * wavparse->rate / GST_SECOND) * wavparse->blockalign;
break;
case GST_FORMAT_DEFAULT:
*dest_value = 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
("bytes_per_sample 0, probably an mp3 - channels %d, width %d",
wavparse->channels, wavparse->width);
res = FALSE;
goto done;
}
no_bps:
{
g_warning ("bps is 0, internal error\n");
res = FALSE;
goto done;
}
}
static const GstQueryType *
gst_wavparse_get_query_types (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_POSITION,
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 = FALSE;
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 ("event %d", GST_EVENT_TYPE (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType start_type, stop_type;
gint64 start, stop;
GstFormat bformat = GST_FORMAT_BYTES;
gint64 bstart, bstop;
gboolean update_start = TRUE;
gboolean update_stop = TRUE;
gst_event_parse_seek (event, &rate, &format, &flags,
&start_type, &start, &stop_type, &stop);
GST_DEBUG ("seek format %d", format);
/* find the corresponding byte position */
if (format != GST_FORMAT_BYTES) {
res &= gst_wavparse_pad_convert (pad, format, start, &bformat, &bstart);
res &= gst_wavparse_pad_convert (pad, format, stop, &bformat, &bstop);
if (!res)
goto done;
}
switch (start_type) {
case GST_SEEK_TYPE_CUR:
bstart = wavparse->segment_start + bstart;
break;
case GST_SEEK_TYPE_END:
bstart = wavparse->datasize + bstart;
break;
case GST_SEEK_TYPE_NONE:
bstart = wavparse->segment_start;
update_start = FALSE;
break;
case GST_SEEK_TYPE_SET:
break;
}
bstart = CLAMP (bstart, 0, wavparse->datasize);
switch (stop_type) {
case GST_SEEK_TYPE_CUR:
bstop = wavparse->segment_stop + bstop;
break;
case GST_SEEK_TYPE_END:
bstop = wavparse->datasize + bstop;
break;
case GST_SEEK_TYPE_NONE:
bstop = wavparse->segment_stop;
update_stop = FALSE;
break;
case GST_SEEK_TYPE_SET:
break;
}
bstop = CLAMP (bstop, 0, wavparse->datasize);
/* now store the values */
wavparse->segment_rate = rate;
wavparse->segment_flags = flags;
wavparse->segment_start = bstart;
wavparse->segment_stop = bstop;
gst_wavparse_handle_seek (wavparse, update_stop || update_start);
break;
}
default:
res = FALSE;
break;
}
done:
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);
GST_DEBUG ("pull_range not supported on sinkpad");
return FALSE;
};
static gboolean
gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
if (active) {
/* if we have a scheduler we can start the task */
gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop, sinkpad);
} else {
gst_pad_stop_task (sinkpad);
}
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:
wav->state = GST_WAVPARSE_START;
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:
gst_wavparse_destroy_sourcepad (wav);
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)
Reference in a new issue View git blame Copy permalink