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gstreamer/gst/wavparse/gstwavparse.c
Stefan Kost 497d589d56 gst/auparse/gstauparse.c: limit caps to the formats we announce in the template 
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
limit caps to the formats we announce in the template
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
fix some crashers/asserts when dealing with broken files
2007-04-10 12:01:33 +00:00

2064 lines
59 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>
* Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
*
* 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>
* Wavparse supports both push and pull mode operations, making it possible to
* stream from a network source.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc location=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>
* <para>
* <programlisting>
* gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* </programlisting>
* Stream data from a network url.
* </para>
* </refsect2>
*
* Last reviewed on 2007-02-14 (0.10.6)
*/
/*
* TODO:
* http://replaygain.hydrogenaudio.org/file_format_wav.html
*/
#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_dispose (GObject * object);
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 const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
static gboolean gst_wavparse_pad_convert (GstPad * pad,
GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
static void gst_wavparse_loop (GstPad * pad);
static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
static const GstElementDetails gst_wavparse_details =
GST_ELEMENT_DETAILS ("WAV audio demuxer",
"Codec/Demuxer/Audio",
"Parse a .wav file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
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, "
"width = (int) { 16, 24, 32 }, "
"depth = (int) [ 1, 32 ], "
"rate = (int) [ 8000, 96000 ], "
"channels = (int) [ 1, 8 ]; "
"audio/x-raw-int, "
"endianness = (int) little_endian, "
"signed = (boolean) false, "
"width = (int) 8, "
"depth = (int) [ 1, 8 ], "
"rate = (int) [ 8000, 96000 ], "
"channels = (int) [ 1, 8 ]; "
"audio/x-raw-float, "
"width = (int) { 32, 64 }, "
"endianness = (int) little_endian, "
"rate = (int) [ 8000, 96000 ], "
"channels = (int) [ 1, 8 ]; "
"audio/ms-gsm; "
"audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int) [ 1, 3 ], "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/mpeg, "
"mpegversion = (int) 4, "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 8 ]; "
"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 ]")
);
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
GST_BOILERPLATE_FULL (GstWavParse, gst_wavparse, GstElement,
GST_TYPE_ELEMENT, DEBUG_INIT);
static void
gst_wavparse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
/* 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));
gst_element_class_set_details (element_class, &gst_wavparse_details);
}
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->dispose = gst_wavparse_dispose;
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
}
static void
gst_wavparse_dispose (GObject * object)
{
GstWavParse *wav;
GST_DEBUG ("WAV: Dispose");
wav = GST_WAVPARSE (object);
if (wav->adapter) {
g_object_unref (wav->adapter);
wav->adapter = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
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->fact = 0;
wavparse->offset = 0;
wavparse->end_offset = 0;
wavparse->dataleft = 0;
wavparse->datasize = 0;
wavparse->datastart = 0;
wavparse->got_fmt = FALSE;
wavparse->first = TRUE;
if (wavparse->seek_event)
gst_event_unref (wavparse->seek_event);
wavparse->seek_event = NULL;
if (wavparse->adapter)
gst_adapter_clear (wavparse->adapter);
}
static void
gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class)
{
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_pad_set_chain_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_chain));
gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
/* src, will be created later */
wavparse->srcpad = NULL;
}
static void
gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
{
if (wavparse->srcpad) {
gst_element_remove_pad (GST_ELEMENT_CAST (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));
GST_DEBUG_OBJECT (wavparse, "srcpad created");
}
/* Compute (value * nom) % denom, avoiding overflow. This can be used
* to perform ceiling or rounding division together with
* gst_util_uint64_scale[_int]. */
#define uint64_scale_modulo(val, nom, denom) \
((val % denom) * (nom % denom) % denom)
/* Like gst_util_uint64_scale_int, but performs ceiling division. */
static guint64
uint64_ceiling_scale_int (guint64 val, gint num, gint denom)
{
guint64 result = gst_util_uint64_scale_int (val, num, denom);
if (uint64_scale_modulo (val, num, denom) == 0)
return result;
else
return result + 1;
}
#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");
}
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");
}
/* 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))
goto no_fmt;
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_bps;
/* Note: gst_riff_create_audio_caps might need 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 == NULL)
goto no_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_CAST (wav), wav->srcpad);
gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
return TRUE;
/* ERRORS */
no_fmt:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
("No FMT tag found"));
return FALSE;
}
no_channels:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to contain zero channels - invalid data"));
g_free (header);
return FALSE;
}
no_bps:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to bitrate of <= zero - invalid data"));
g_free (header);
return FALSE;
}
no_caps:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
return FALSE;
}
}
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;
GST_DEBUG_OBJECT (wav, "datasize = %ld", 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_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 a 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_CAST (wav), buf))
return GST_FLOW_ERROR;
wav->offset += 12;
return GST_FLOW_OK;
}
/* 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;
GstFormat format, bformat;
GstSeekFlags flags;
GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
gint64 cur, stop, upstream_size;
gboolean flush;
gboolean update;
GstSegment seeksegment = { 0, };
if (event) {
GstFormat fmt;
GST_DEBUG_OBJECT (wav, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
/* no negative rates yet */
if (rate < 0.0)
goto negative_rate;
fmt = wav->segment.format;
/* we have to have a format as the segment format. Try to convert
* if not. */
if (format != wav->segment.format) {
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;
cur_type = GST_SEEK_TYPE_SET;
stop_type = GST_SEEK_TYPE_SET;
}
/* get flush flag */
flush = flags & GST_SEEK_FLAG_FLUSH;
/* now we need to make sure the streaming thread is stopped. We do this by
* either sending a FLUSH_START event downstream which will cause the
* streaming thread to stop with a WRONG_STATE.
* For a non-flushing seek we simply pause the task, which will happen as soon
* as it completes one iteration (and thus might block when the sink is
* blocking in preroll). */
if (flush) {
if (wav->srcpad) {
GST_DEBUG_OBJECT (wav, "sending flush start");
gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
}
} else {
gst_pad_pause_task (wav->sinkpad);
}
/* we should now be able to grab the streaming thread because we stopped it
* with the above flush/pause code */
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));
/* configure the seek parameters in the seeksegment. We will then have the
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (wav, "configuring seek");
gst_segment_set_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
if ((cur_type != GST_SEEK_TYPE_NONE)) {
/* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
* we can just copy the last_stop. If not, we use the bps to convert TIME to
* bytes. */
if (wav->bps)
wav->offset =
uint64_ceiling_scale_int (seeksegment.last_stop, wav->bps,
GST_SECOND);
else
wav->offset = seeksegment.last_stop;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset -= (wav->offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset += wav->datastart;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
} else {
GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
wav->offset);
}
if (stop_type != GST_SEEK_TYPE_NONE) {
if (wav->bps)
wav->end_offset = uint64_ceiling_scale_int (stop, wav->bps, GST_SECOND);
else
wav->end_offset = stop;
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
wav->end_offset += wav->datastart;
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
} else {
GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
wav->end_offset);
}
/* make sure filesize is not exceeded due to rounding errors or so,
* same precaution as in _stream_headers */
bformat = GST_FORMAT_BYTES;
if (gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size))
wav->end_offset = MIN (wav->end_offset, upstream_size);
/* this is the range of bytes we will use for playback */
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 (wav->srcpad) {
if (flush) {
/* if we sent a FLUSH_START, we now send a FLUSH_STOP */
GST_DEBUG_OBJECT (wav, "sending flush stop");
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 previous last_stop. */
GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.accum, wav->segment.last_stop);
gst_pad_push_event (wav->srcpad,
gst_event_new_new_segment (TRUE,
wav->segment.rate, wav->segment.format,
wav->segment.accum, wav->segment.last_stop, wav->segment.accum));
/* keep track of our last_stop */
seeksegment.accum = wav->segment.last_stop;
}
}
/* now we did the seek and can activate the new segment values */
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
/* if we're doing a segment seek, post a SEGMENT_START message */
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (wav),
gst_message_new_segment_start (GST_OBJECT_CAST (wav),
wav->segment.format, wav->segment.last_stop));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.last_stop, stop);
/* store the newsegment event so it can be sent from the streaming thread. */
if (wav->newsegment)
gst_event_unref (wav->newsegment);
wav->newsegment =
gst_event_new_new_segment (FALSE, wav->segment.rate,
wav->segment.format, wav->segment.last_stop, stop,
wav->segment.last_stop);
/* and start the streaming task again */
wav->segment_running = TRUE;
if (!wav->streaming) {
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
wav->sinkpad);
}
GST_PAD_STREAM_UNLOCK (wav->sinkpad);
return TRUE;
/* ERRORS */
negative_rate:
{
GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
return FALSE;
}
no_format:
{
GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
return FALSE;
}
}
/*
* gst_wavparse_peek_chunk_info:
* @wav Wavparse object
* @tag holder for tag
* @size holder for tag size
*
* Peek next chunk info (tag and size)
*
* Returns: %TRUE when one chunk info has been got from the adapter
*/
static gboolean
gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
{
const guint8 *data = NULL;
if (gst_adapter_available (wav->adapter) < 8)
return FALSE;
data = gst_adapter_peek (wav->adapter, 8);
*tag = GST_READ_UINT32_LE (data);
*size = GST_READ_UINT32_LE (data + 4);
GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size,
GST_FOURCC_ARGS (*tag));
return TRUE;
}
/*
* gst_wavparse_peek_chunk:
* @wav Wavparse object
* @tag holder for tag
* @size holder for tag size
*
* Peek enough data for one full chunk
*
* Returns: %TRUE when one chunk has been got
*/
static gboolean
gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
{
guint32 peek_size = 0;
guint available;
if (!gst_wavparse_peek_chunk_info (wav, tag, size))
return FALSE;
GST_DEBUG ("Need to peek chunk of %d bytes", *size);
peek_size = (*size + 1) & ~1;
available = gst_adapter_available (wav->adapter);
if (available >= (8 + peek_size)) {
return TRUE;
} else {
GST_LOG ("but only %u bytes available now", available);
return FALSE;
}
}
static gboolean
gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len)
{
gboolean res = FALSE;
GstFormat fmt = GST_FORMAT_BYTES;
GstPad *peer;
if ((peer = gst_pad_get_peer (wav->sinkpad))) {
res = gst_pad_query_duration (peer, &fmt, len);
gst_object_unref (peer);
}
return res;
}
static GstFlowReturn
gst_wavparse_stream_headers (GstWavParse * wav)
{
GstFlowReturn res;
GstBuffer *buf;
gst_riff_strf_auds *header = NULL;
guint32 tag, size;
gboolean gotdata = FALSE;
GstCaps *caps;
gint64 duration;
gchar *codec_name = NULL;
GstEvent **event_p;
while (!wav->got_fmt) {
GstBuffer *extra;
/* The header starts with a 'fmt ' tag */
if (wav->streaming) {
if (!gst_wavparse_peek_chunk (wav, &tag, &size))
return GST_FLOW_OK;
gst_adapter_flush (wav->adapter, 8);
wav->offset += 8;
buf = gst_adapter_take_buffer (wav->adapter, size);
} else {
if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
&wav->offset, &tag, &buf)) != GST_FLOW_OK)
return res;
}
if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_bext ||
tag == GST_RIFF_TAG_BEXT) {
GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
GST_FOURCC_ARGS (tag));
if (wav->streaming) {
gst_adapter_flush (wav->adapter, size);
wav->offset += size;
}
gst_buffer_unref (buf);
buf = NULL;
continue;
}
if (tag != GST_RIFF_TAG_fmt)
goto invalid_wav;
if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
&extra)))
goto parse_header_error;
buf = NULL; /* parse_strf_auds() took ownership of buffer */
if (header->channels == 0)
goto no_channels;
GST_DEBUG_OBJECT (wav, "creating the caps");
/* Note: gst_riff_create_audio_caps might need 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);
if (!caps)
goto unknown_format;
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
wav->blockalign = header->blockalign;
wav->depth = header->size;
wav->av_bps = header->av_bps;
g_free (header);
/* do format specific handling */
switch (wav->format) {
case GST_RIFF_WAVE_FORMAT_MPEGL12:
case GST_RIFF_WAVE_FORMAT_MPEGL3:
{
/* Note: workaround for mp2/mp3 embedded in wav, that relies on the
* bitrate inside the mpeg stream */
GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps);
wav->bps = 0;
break;
}
default:
/* use the configured bps */
wav->bps = wav->av_bps;
break;
}
wav->width = (wav->blockalign * 8) / wav->channels;
wav->bytes_per_sample = wav->channels * wav->width / 8;
if (wav->bytes_per_sample <= 0)
goto no_bytes_per_sample;
GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
GST_DEBUG_OBJECT (wav, "frequency = %d", wav->rate);
GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
/* bps can be 0 when we don't have a valid bitrate (mostly for compressed
* formats). This will make the element output a BYTE format segment and
* will not timestamp the outgoing buffers.
*/
GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
/* 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);
wav->got_fmt = TRUE;
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;
}
}
/* loop headers until we get data */
while (!gotdata) {
gint64 upstream_size = 0;
if (wav->streaming) {
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
return GST_FLOW_OK;
} else {
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
&buf)) != GST_FLOW_OK)
goto header_read_error;
tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
}
gst_wavparse_get_upstream_size (wav, &upstream_size);
/* 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
*/
switch (tag) {
/* TODO : Implement the various cases */
case GST_RIFF_TAG_data:{
GstFormat fmt;
GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
if (wav->streaming) {
gst_adapter_flush (wav->adapter, 8);
gotdata = TRUE;
} else {
gst_buffer_unref (buf);
}
wav->offset += 8;
wav->datastart = wav->offset;
/* file might be truncated */
fmt = GST_FORMAT_BYTES;
if (upstream_size) {
size = MIN (size, (upstream_size - wav->datastart));
}
wav->datasize = (guint64) size;
wav->dataleft = (guint64) size;
wav->end_offset = size + wav->datastart;
if (!wav->streaming) {
/* We will continue parsing tags 'till end */
wav->offset += size;
}
GST_DEBUG_OBJECT (wav, "datasize = %d", size);
break;
}
case GST_RIFF_TAG_fact:{
/* number of samples (for compressed formats) */
if (wav->streaming) {
const guint8 *data = NULL;
if (gst_adapter_available (wav->adapter) < 8 + 4) {
return GST_FLOW_OK;
}
gst_adapter_flush (wav->adapter, 8);
data = gst_adapter_peek (wav->adapter, 4);
wav->fact = GST_READ_UINT32_LE (data);
gst_adapter_flush (wav->adapter, 4);
} else {
gst_buffer_unref (buf);
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8, 4,
&buf)) != GST_FLOW_OK)
goto header_read_error;
wav->fact = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
gst_buffer_unref (buf);
}
wav->offset += 8 + 4;
break;
}
default:
if (wav->streaming) {
if (!gst_wavparse_peek_chunk (wav, &tag, &size))
return GST_FLOW_OK;
}
GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (tag));
wav->offset += 8 + ((size + 1) & ~1);
if (wav->streaming) {
gst_adapter_flush (wav->adapter, 8 + ((size + 1) & ~1));
} else {
gst_buffer_unref (buf);
}
}
if (upstream_size && (wav->offset >= upstream_size)) {
/* Now we are gone through the whole file */
gotdata = TRUE;
}
}
GST_DEBUG_OBJECT (wav, "Finished parsing headers");
if (wav->bps <= 0 && wav->fact) {
wav->bps =
(guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
(guint64) wav->fact);
GST_DEBUG_OBJECT (wav, "calculated bps : %d", wav->bps);
}
if (wav->bps > 0) {
duration = uint64_ceiling_scale_int (wav->datasize, GST_SECOND, wav->bps);
GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
gst_segment_init (&wav->segment, GST_FORMAT_TIME);
gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, duration);
} else {
/* no bitrate, let downstream peer do the math, we'll feed it bytes. */
gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
gst_segment_set_duration (&wav->segment, GST_FORMAT_BYTES, wav->datasize);
}
/* 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);
wav->state = GST_WAVPARSE_DATA;
return GST_FLOW_OK;
/* ERROR */
invalid_wav:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (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"));
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_bytes_per_sample:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("could not caluclate bytes per sample - invalid data"));
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 (header);
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;
}
}
/*
* Read WAV file tag when streaming
*/
static GstFlowReturn
gst_wavparse_parse_stream_init (GstWavParse * wav)
{
if (gst_adapter_available (wav->adapter) >= 12) {
GstBuffer *tmp;
/* _take flushes the data */
tmp = gst_adapter_take_buffer (wav->adapter, 12);
GST_DEBUG ("Parsing wav header");
if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
return GST_FLOW_ERROR;
wav->offset += 12;
/* Go to next state */
wav->state = GST_WAVPARSE_HEADER;
}
return GST_FLOW_OK;
}
/* 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;
GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
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 };
GST_DEBUG_OBJECT (wav, "adding src pad");
if (wav->caps) {
s = gst_caps_get_structure (wav->caps, 0);
if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf &&
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_CAST (wav), wav->srcpad);
gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
GST_DEBUG_OBJECT (wav, "Send newsegment event on newpad");
if (wav->newsegment) {
gst_pad_push_event (wav->srcpad, wav->newsegment);
wav->newsegment = NULL;
}
if (wav->tags) {
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (wav), wav->srcpad,
wav->tags);
wav->tags = NULL;
}
}
#define MAX_BUFFER_SIZE 4096
static GstFlowReturn
gst_wavparse_stream_data (GstWavParse * wav)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
GstClockTime timestamp, next_timestamp, duration;
guint64 pos, nextpos;
iterate_adapter:
GST_LOG_OBJECT (wav,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
/* Get the next n bytes and output them */
if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
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 (gst_guint64_to_gdouble (wav->dataleft),
MAX_BUFFER_SIZE * wav->segment.abs_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 (wav->streaming) {
guint avail = gst_adapter_available (wav->adapter);
if (avail < desired) {
GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
return GST_FLOW_OK;
}
buf = gst_adapter_take_buffer (wav->adapter, desired);
} else {
if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
desired, &buf)) != GST_FLOW_OK)
goto pull_error;
}
/* 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 (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
gst_wavparse_add_src_pad (wav, buf);
}
/* If we have a pending newsegment send it now. */
if (G_UNLIKELY (wav->newsegment != NULL)) {
gst_pad_push_event (wav->srcpad, wav->newsegment);
wav->newsegment = NULL;
}
obtained = GST_BUFFER_SIZE (buf);
/* our positions in bytes */
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;
if (wav->bps > 0) {
/* and timestamps if we have a bitrate, be carefull for overflows */
timestamp = uint64_ceiling_scale_int (pos, GST_SECOND, wav->bps);
next_timestamp = uint64_ceiling_scale_int (nextpos, GST_SECOND, wav->bps);
duration = next_timestamp - timestamp;
/* update current running segment position */
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp);
} else {
/* no bitrate, don't timestamp */
timestamp = GST_CLOCK_TIME_NONE;
next_timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_BYTES, nextpos);
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
/* don't forget to set the caps on the buffer */
gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad));
GST_LOG_OBJECT (wav,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
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;
} else {
wav->dataleft = 0;
}
wav->offset += obtained;
/* Iterate until need more data, so adapter size won't grow */
if (wav->streaming) {
GST_LOG_OBJECT (wav,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
wav->end_offset);
goto iterate_adapter;
}
return res;
/* ERROR */
found_eos:
{
GST_DEBUG_OBJECT (wav, "found EOS");
return GST_FLOW_UNEXPECTED;
}
pull_error:
{
/* check if we got EOS */
if (res == GST_FLOW_UNEXPECTED)
goto found_eos;
GST_WARNING_OBJECT (wav,
"Error getting %" G_GINT64_FORMAT " bytes from the "
"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
return res;
}
push_error:
{
GST_WARNING_OBJECT (wav, "Error pushing on srcpad %p, is linked? = %d",
wav->srcpad, gst_pad_is_linked (wav->srcpad));
return res;
}
}
static void
gst_wavparse_loop (GstPad * pad)
{
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (wav, "process data");
switch (wav->state) {
case GST_WAVPARSE_START:
GST_DEBUG_OBJECT (wav, "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:
GST_DEBUG_OBJECT (wav, "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 ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto pause;
break;
default:
g_assert_not_reached ();
}
return;
/* ERRORS */
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
wav->segment_running = FALSE;
gst_pad_pause_task (pad);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
if (ret == GST_FLOW_UNEXPECTED) {
/* add pad before we perform EOS */
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
gst_wavparse_add_src_pad (wav, NULL);
}
/* perform EOS logic */
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_CAST (wav),
gst_message_new_segment_done (GST_OBJECT_CAST (wav),
wav->segment.format, stop));
} else {
if (wav->srcpad != NULL)
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
} else {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (wav, STREAM, FAILED,
(_("Internal data flow error.")),
("streaming task paused, reason %s (%d)", reason, ret));
if (wav->srcpad != NULL)
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
}
return;
}
}
static GstFlowReturn
gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (wav, "adapter_push %u bytes", GST_BUFFER_SIZE (buf));
gst_adapter_push (wav->adapter, buf);
switch (wav->state) {
case GST_WAVPARSE_START:
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
goto done;
if (wav->state != GST_WAVPARSE_HEADER)
break;
/* otherwise fall-through */
case GST_WAVPARSE_HEADER:
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto done;
if (!wav->got_fmt || wav->datastart == 0)
break;
wav->state = GST_WAVPARSE_DATA;
/* fall-through */
case GST_WAVPARSE_DATA:
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto done;
break;
default:
g_assert_not_reached ();
}
done:
return ret;
}
#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 (*dest_format == src_format) {
*dest_value = src_value;
return TRUE;
}
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;
/* make sure we end up on a sample boundary */
*dest_value -= *dest_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;
}
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_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;
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;
endb = wav->datasize;
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:{
if (wav->fact) {
end = GST_SECOND * wav->fact / wav->rate;
} else {
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, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (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);
gst_event_unref (event);
break;
}
default:
res = gst_pad_push_event (wavparse->sinkpad, event);
break;
}
return res;
}
static gboolean
gst_wavparse_sink_activate (GstPad * sinkpad)
{
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
gboolean res;
if (wav->adapter)
gst_object_unref (wav->adapter);
if (gst_pad_check_pull_range (sinkpad)) {
GST_DEBUG ("going to pull mode");
wav->streaming = FALSE;
wav->adapter = NULL;
res = gst_pad_activate_pull (sinkpad, TRUE);
} else {
GST_DEBUG ("going to push (streaming) mode");
wav->streaming = TRUE;
wav->adapter = gst_adapter_new ();
res = gst_pad_activate_push (sinkpad, TRUE);
}
gst_object_unref (wav);
return res;
}
static gboolean
gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
GST_DEBUG_OBJECT (wav, "activating pull");
if (active) {
/* if we have a scheduler we can start the task */
wav->segment_running = TRUE;
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:
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)
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