1 /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */

     2 /* GStreamer

     3  * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>

     4  * Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.

     5  *

     6  * This library is free software; you can redistribute it and/or

     7  * modify it under the terms of the GNU Library General Public

     8  * License as published by the Free Software Foundation; either

     9  * version 2 of the License, or (at your option) any later version.

    10  *

    11  * This library is distributed in the hope that it will be useful,

    12  * but WITHOUT ANY WARRANTY; without even the implied warranty of

    13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

    14  * Library General Public License for more details.

    15  *

    16  * You should have received a copy of the GNU Library General Public

    17  * License along with this library; if not, write to the

    18  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,

    19  * Boston, MA 02111-1307, USA.

    20  */

    21 

    22 /**

    23  * SECTION:element-wavparse

    24  *

    25  * <refsect2>

    26  * <para>

    27  * Parse a .wav file into raw or compressed audio.

    28  * </para>

    29  * <para>

    30  * This element currently only supports pull based scheduling.

    31  * </para>

    32  * <title>Example launch line</title>

    33  * <para>

    34  * <programlisting>

    35  * gst-launch filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink

    36  * </programlisting>

    37  * Read a wav file and output to the soundcard using the ALSA element. The

    38  * wav file is assumed to contain raw uncompressed samples.

    39  * </para>

    40  * <para>

    41  * <programlisting>

    42  * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink

    43  * </programlisting>

    44  * Stream data from

    45  * </para>

    46  * </refsect2>

    47  *

    48  * Last reviewed on 2006-03-03 (0.10.3)

    49  */

    50 

    51 #ifdef HAVE_CONFIG_H

    52 #include "config.h"

    53 #endif

    54 

    55 #include "string.h"

    56 #include "gstwavparse.h"

    57 #include <gst/riff/riff-ids.h>

    58 #include <gst/riff/riff-media.h>

    59 #include <gst/riff/riff-read.h>

    60 

    61 #ifndef __SYMBIAN32__

    62 #include <gst/gst-i18n-plugin.h>

    63 #else

    64 #include "gst/gst-i18n-plugin.h"

    65 #endif

    66 

    67 #ifdef __SYMBIAN32__

    68 #include <gst/gstinfo.h>

    69 #endif

    70 

    71 #ifndef G_MAXUINT32

    72 #define G_MAXUINT32 0xffffffff

    73 #endif

    74 

    75 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);

    76 #define GST_CAT_DEFAULT (wavparse_debug)

    77 

    78 static void gst_wavparse_base_init (gpointer g_class);

    79 static void gst_wavparse_class_init (GstWavParseClass * klass);

    80 static void gst_wavparse_init (GstWavParse * wavparse);

    81 static void gst_wavparse_dispose (GObject * object);

    82 

    83 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad);

    84 static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,

    85     gboolean active);

    86 static gboolean gst_wavparse_send_event (GstElement * element,

    87     GstEvent * event);

    88 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);

    89 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,

    90     GstStateChange transition);

    91 

    92 static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);

    93 static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);

    94 static gboolean gst_wavparse_pad_convert (GstPad * pad,

    95     GstFormat src_format,

    96     gint64 src_value, GstFormat * dest_format, gint64 * dest_value);

    97 

    98 static void gst_wavparse_loop (GstPad * pad);

    99 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);

   100 static void gst_wavparse_get_property (GObject * object, guint prop_id,

   101     GValue * value, GParamSpec * pspec);

   102 

   103 static const GstElementDetails gst_wavparse_details =

   104 GST_ELEMENT_DETAILS ("WAV audio demuxer",

   105     "Codec/Demuxer/Audio",

   106     "Parse a .wav file into raw audio",

   107     "Erik Walthinsen <omega@cse.ogi.edu>");

   108 

   109 static GstStaticPadTemplate sink_template_factory =

   110 //GST_STATIC_PAD_TEMPLATE ("wavparse_sink",

   111 GST_STATIC_PAD_TEMPLATE ("sink", 

   112     GST_PAD_SINK,

   113     GST_PAD_ALWAYS,

   114     GST_STATIC_CAPS ("audio/x-wav")

   115     );

   116 

   117 /* the pad is marked a sometimes and is added to the element when the

   118  * exact type is known. This makes it much easier for a static autoplugger

   119  * to connect the right decoder when needed.

   120  */

   121 static GstStaticPadTemplate src_template_factory =

   122  //   GST_STATIC_PAD_TEMPLATE ("wavparse_src",

   123     GST_STATIC_PAD_TEMPLATE ("src",

   124     GST_PAD_SRC,

   125     GST_PAD_SOMETIMES,

   126     GST_STATIC_CAPS ("audio/x-raw-int, "

   127         "endianness = (int) little_endian, "

   128         "signed = (boolean) { true, false }, "

   129         "width = (int) { 8, 16, 24, 32 }, "

   130         "depth = (int) { 8, 16, 24, 32 }, "

   131         "rate = (int) [ 8000, 96000 ], "

   132         "channels = (int) [ 1, 8 ]; "

   133         "audio/mpeg, "

   134         "mpegversion = (int) 1, "

   135         "layer = (int) [ 1, 3 ], "

   136         "rate = (int) [ 8000, 48000 ], "

   137         "channels = (int) [ 1, 2 ]; "

   138         "audio/x-alaw, "

   139         "rate = (int) [ 8000, 48000 ], "

   140         "channels = (int) [ 1, 2 ]; "

   141         "audio/x-mulaw, "

   142         "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"

   143         "audio/x-adpcm, "

   144         "layout = (string) microsoft, "

   145         "block_align = (int) [ 1, 8192 ], "

   146         "rate = (int) [ 8000, 48000 ], "

   147         "channels = (int) [ 1, 2 ]; "

   148         "audio/x-adpcm, "

   149         "layout = (string) dvi, "

   150         "block_align = (int) [ 1, 8192 ], "

   151         "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"

   152         "audio/x-vnd.sony.atrac3;"

   153         "audio/x-dts;" "audio/x-wma, " "wmaversion = (int) [ 1, 2 ]")

   154     );

   155 

   156 

   157 static GstElementClass *parent_class = NULL;

   158 

   159 GType

   160 gst_wavparse_get_type (void)

   161 {

   162   static GType wavparse_type = 0;

   163 

   164   if (!wavparse_type) {

   165     static const GTypeInfo wavparse_info = {

   166       sizeof (GstWavParseClass),

   167       gst_wavparse_base_init,

   168       NULL,

   169       (GClassInitFunc) gst_wavparse_class_init,

   170       NULL,

   171       NULL,

   172       sizeof (GstWavParse),

   173       0,

   174       (GInstanceInitFunc) gst_wavparse_init,

   175     };

   176 

   177     wavparse_type =

   178         g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse",

   179         &wavparse_info, 0);

   180   }

   181   return wavparse_type;

   182 }

   183 

   184 

   185 static void

   186 gst_wavparse_base_init (gpointer g_class)

   187 {

   188   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

   189 

   190   /* register src pads */

   191   gst_element_class_add_pad_template (element_class,

   192       gst_static_pad_template_get (&sink_template_factory));

   193   gst_element_class_add_pad_template (element_class,

   194       gst_static_pad_template_get (&src_template_factory));

   195   gst_element_class_set_details (element_class, &gst_wavparse_details);

   196 }

   197 

   198 static void

   199 gst_wavparse_class_init (GstWavParseClass * klass)

   200 {

   201   GstElementClass *gstelement_class;

   202   GObjectClass *object_class;

   203 

   204   gstelement_class = (GstElementClass *) klass;

   205   object_class = (GObjectClass *) klass;

   206 

   207   parent_class = g_type_class_peek_parent (klass);

   208 

   209   object_class->get_property = gst_wavparse_get_property;

   210   object_class->dispose = gst_wavparse_dispose;

   211 

   212   gstelement_class->change_state = gst_wavparse_change_state;

   213   gstelement_class->send_event = gst_wavparse_send_event;

   214 }

   215 

   216 

   217 static void

   218 gst_wavparse_dispose (GObject * object)

   219 {

   220 	#ifndef __SYMBIAN32__

   221   GST_DEBUG("WAV: Dispose\n");

   222   #endif

   223   GstWavParse *wav = GST_WAVPARSE (object);

   224   #ifdef __SYMBIAN32__

   225   GST_DEBUG("WAV: Dispose\n");

   226 	#endif

   227 

   228   if (wav->adapter) {

   229     g_object_unref (wav->adapter);

   230     wav->adapter = NULL;

   231   }

   232 

   233   G_OBJECT_CLASS (parent_class)->dispose (object);

   234 }

   235 

   236 

   237 static void

   238 gst_wavparse_reset (GstWavParse * wavparse)

   239 {

   240   wavparse->state = GST_WAVPARSE_START;

   241 

   242   /* These will all be set correctly in the fmt chunk */

   243   wavparse->depth = 0;

   244   wavparse->rate = 0;

   245   wavparse->width = 0;

   246   wavparse->channels = 0;

   247   wavparse->blockalign = 0;

   248   wavparse->bps = 0;

   249   wavparse->offset = 0;

   250   wavparse->end_offset = 0;

   251   wavparse->dataleft = 0;

   252   wavparse->datasize = 0;

   253   wavparse->datastart = 0;

   254   wavparse->got_fmt = FALSE;

   255   wavparse->first = TRUE;

   256 

   257   if (wavparse->seek_event)

   258     gst_event_unref (wavparse->seek_event);

   259   wavparse->seek_event = NULL;

   260 

   261   /* we keep the segment info in time */

   262   gst_segment_init (&wavparse->segment, GST_FORMAT_TIME);

   263 }

   264 

   265 static void

   266 gst_wavparse_init (GstWavParse * wavparse)

   267 {

   268   gst_wavparse_reset (wavparse);

   269 

   270   /* sink */

   271   wavparse->sinkpad =

   272       gst_pad_new_from_static_template (&sink_template_factory, "sink");

   273   gst_pad_set_activate_function (wavparse->sinkpad,

   274       GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));

   275   gst_pad_set_activatepull_function (wavparse->sinkpad,

   276       GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));

   277   gst_pad_set_chain_function (wavparse->sinkpad,

   278       GST_DEBUG_FUNCPTR (gst_wavparse_chain));

   279   gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);

   280 

   281   /* src, will be created later */

   282   wavparse->srcpad = NULL;

   283 }

   284 

   285 static void

   286 gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)

   287 {

   288   if (wavparse->srcpad) {

   289     gst_element_remove_pad (GST_ELEMENT (wavparse), wavparse->srcpad);

   290     wavparse->srcpad = NULL;

   291   }

   292 }

   293 

   294 static void

   295 gst_wavparse_create_sourcepad (GstWavParse * wavparse)

   296 {

   297   /* destroy previous one */

   298   gst_wavparse_destroy_sourcepad (wavparse);

   299 

   300   /* source */

   301   wavparse->srcpad =

   302       gst_pad_new_from_static_template (&src_template_factory, "src");

   303   gst_pad_use_fixed_caps (wavparse->srcpad);

   304   gst_pad_set_query_type_function (wavparse->srcpad,

   305       GST_DEBUG_FUNCPTR (gst_wavparse_get_query_types));

   306   gst_pad_set_query_function (wavparse->srcpad,

   307       GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));

   308   gst_pad_set_event_function (wavparse->srcpad,

   309       GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));

   310 

   311   GST_DEBUG_OBJECT (wavparse, "srcpad created");

   312 }

   313 

   314 static void

   315 gst_wavparse_get_property (GObject * object,

   316     guint prop_id, GValue * value, GParamSpec * pspec)

   317 {

   318   GstWavParse *wavparse;

   319 

   320   wavparse = GST_WAVPARSE (object);

   321 

   322   switch (prop_id) {

   323     default:

   324       break;

   325   }

   326 }

   327 

   328 

   329 

   330 #if 0

   331 static void

   332 gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)

   333 {

   334   guint32 got_bytes;

   335   GstByteStream *bs = wavparse->bs;

   336   gst_riff_chunk *temp_chunk, chunk;

   337   guint8 *tempdata;

   338   struct _gst_riff_labl labl, *temp_labl;

   339   struct _gst_riff_ltxt ltxt, *temp_ltxt;

   340   struct _gst_riff_note note, *temp_note;

   341   char *label_name;

   342   GstProps *props;

   343   GstPropsEntry *entry;

   344   GstCaps *new_caps;

   345   GList *caps = NULL;

   346 

   347   props = wavparse->metadata->properties;

   348 

   349   while (len > 0) {

   350     got_bytes =

   351         gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));

   352     if (got_bytes != sizeof (gst_riff_chunk)) {

   353       return;

   354     }

   355     temp_chunk = (gst_riff_chunk *) tempdata;

   356 

   357     chunk.id = GUINT32_FROM_LE (temp_chunk->id);

   358     chunk.size = GUINT32_FROM_LE (temp_chunk->size);

   359 

   360     if (chunk.size == 0) {

   361       gst_bytestream_flush (bs, sizeof (gst_riff_chunk));

   362       len -= sizeof (gst_riff_chunk);

   363       continue;

   364     }

   365 

   366     switch (chunk.id) {

   367       case GST_RIFF_adtl_labl:

   368         got_bytes =

   369             gst_bytestream_peek_bytes (bs, &tempdata,

   370             sizeof (struct _gst_riff_labl));

   371         if (got_bytes != sizeof (struct _gst_riff_labl)) {

   372           return;

   373         }

   374 

   375         temp_labl = (struct _gst_riff_labl *) tempdata;

   376         labl.id = GUINT32_FROM_LE (temp_labl->id);

   377         labl.size = GUINT32_FROM_LE (temp_labl->size);

   378         labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);

   379 

   380         gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));

   381         len -= sizeof (struct _gst_riff_labl);

   382 

   383         got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);

   384         if (got_bytes != labl.size - 4) {

   385           return;

   386         }

   387 

   388         label_name = (char *) tempdata;

   389 

   390         gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);

   391         len -= (((labl.size - 4) + 1) & ~1);

   392 

   393         new_caps = gst_caps_new ("label",

   394             "application/x-gst-metadata",

   395             gst_props_new ("identifier", G_TYPE_INT (labl.identifier),

   396                 "name", G_TYPE_STRING (label_name), NULL));

   397 

   398         if (gst_props_get (props, "labels", &caps, NULL)) {

   399           caps = g_list_append (caps, new_caps);

   400         } else {

   401           caps = g_list_append (NULL, new_caps);

   402 

   403           entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));

   404           gst_props_add_entry (props, entry);

   405         }

   406 

   407         break;

   408 

   409       case GST_RIFF_adtl_ltxt:

   410         got_bytes =

   411             gst_bytestream_peek_bytes (bs, &tempdata,

   412             sizeof (struct _gst_riff_ltxt));

   413         if (got_bytes != sizeof (struct _gst_riff_ltxt)) {

   414           return;

   415         }

   416 

   417         temp_ltxt = (struct _gst_riff_ltxt *) tempdata;

   418         ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);

   419         ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);

   420         ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);

   421         ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);

   422         ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);

   423         ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);

   424         ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);

   425         ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);

   426         ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);

   427 

   428         gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));

   429         len -= sizeof (struct _gst_riff_ltxt);

   430 

   431         if (ltxt.size - 20 > 0) {

   432           got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);

   433           if (got_bytes != ltxt.size - 20) {

   434             return;

   435           }

   436 

   437           gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);

   438           len -= (((ltxt.size - 20) + 1) & ~1);

   439 

   440           label_name = (char *) tempdata;

   441         } else {

   442           label_name = "";

   443         }

   444 

   445         new_caps = gst_caps_new ("ltxt",

   446             "application/x-gst-metadata",

   447             gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),

   448                 "name", G_TYPE_STRING (label_name),

   449                 "length", G_TYPE_INT (ltxt.length), NULL));

   450 

   451         if (gst_props_get (props, "ltxts", &caps, NULL)) {

   452           caps = g_list_append (caps, new_caps);

   453         } else {

   454           caps = g_list_append (NULL, new_caps);

   455 

   456           entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));

   457           gst_props_add_entry (props, entry);

   458         }

   459 

   460         break;

   461 

   462       case GST_RIFF_adtl_note:

   463         got_bytes =

   464             gst_bytestream_peek_bytes (bs, &tempdata,

   465             sizeof (struct _gst_riff_note));

   466         if (got_bytes != sizeof (struct _gst_riff_note)) {

   467           return;

   468         }

   469 

   470         temp_note = (struct _gst_riff_note *) tempdata;

   471         note.id = GUINT32_FROM_LE (temp_note->id);

   472         note.size = GUINT32_FROM_LE (temp_note->size);

   473         note.identifier = GUINT32_FROM_LE (temp_note->identifier);

   474 

   475         gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));

   476         len -= sizeof (struct _gst_riff_note);

   477 

   478         got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);

   479         if (got_bytes != note.size - 4) {

   480           return;

   481         }

   482 

   483         gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);

   484         len -= (((note.size - 4) + 1) & ~1);

   485 

   486         label_name = (char *) tempdata;

   487 

   488         new_caps = gst_caps_new ("note",

   489             "application/x-gst-metadata",

   490             gst_props_new ("identifier", G_TYPE_INT (note.identifier),

   491                 "name", G_TYPE_STRING (label_name), NULL));

   492 

   493         if (gst_props_get (props, "notes", &caps, NULL)) {

   494           caps = g_list_append (caps, new_caps);

   495         } else {

   496           caps = g_list_append (NULL, new_caps);

   497 

   498           entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));

   499           gst_props_add_entry (props, entry);

   500         }

   501 

   502         break;

   503 

   504       default:

   505         g_print ("Unknown chunk: %" GST_FOURCC_FORMAT "\n",

   506             GST_FOURCC_ARGS (chunk.id));

   507         return;

   508     }

   509   }

   510 

   511   g_object_notify (G_OBJECT (wavparse), "metadata");

   512 }

   513 

   514 static void

   515 gst_wavparse_parse_cues (GstWavParse * wavparse, int len)

   516 {

   517   guint32 got_bytes;

   518   GstByteStream *bs = wavparse->bs;

   519   struct _gst_riff_cue *temp_cue, cue;

   520   struct _gst_riff_cuepoints *points;

   521   guint8 *tempdata;

   522   int i;

   523   GList *cues = NULL;

   524   GstPropsEntry *entry;

   525 

   526   while (len > 0) {

   527     int required;

   528 

   529     got_bytes =

   530         gst_bytestream_peek_bytes (bs, &tempdata,

   531         sizeof (struct _gst_riff_cue));

   532     temp_cue = (struct _gst_riff_cue *) tempdata;

   533 

   534     /* fixup for our big endian friends */

   535     cue.id = GUINT32_FROM_LE (temp_cue->id);

   536     cue.size = GUINT32_FROM_LE (temp_cue->size);

   537     cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);

   538 

   539     gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));

   540     if (got_bytes != sizeof (struct _gst_riff_cue)) {

   541       return;

   542     }

   543 

   544     len -= sizeof (struct _gst_riff_cue);

   545 

   546     /* -4 because cue.size contains the cuepoints size

   547        and we've already flushed that out of the system */

   548     required = cue.size - 4;

   549     got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);

   550     gst_bytestream_flush (bs, ((required) + 1) & ~1);

   551     if (got_bytes != required) {

   552       return;

   553     }

   554 

   555     len -= (((cue.size - 4) + 1) & ~1);

   556 

   557     /* now we have an array of struct _gst_riff_cuepoints in tempdata */

   558     points = (struct _gst_riff_cuepoints *) tempdata;

   559 

   560     for (i = 0; i < cue.cuepoints; i++) {

   561       GstCaps *caps;

   562 

   563       caps = gst_caps_new ("cues",

   564           "application/x-gst-metadata",

   565           gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),

   566               "position", G_TYPE_INT (points[i].offset), NULL));

   567       cues = g_list_append (cues, caps);

   568     }

   569 

   570     entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));

   571     gst_props_add_entry (wavparse->metadata->properties, entry);

   572   }

   573 

   574   g_object_notify (G_OBJECT (wavparse), "metadata");

   575 }

   576 

   577 /* Read 'fmt ' header */

   578 static gboolean

   579 gst_wavparse_fmt (GstWavParse * wav)

   580 {

   581   gst_riff_strf_auds *header = NULL;

   582   GstCaps *caps;

   583 

   584   if (!gst_riff_read_strf_auds (wav, &header)) {

   585     g_warning ("Not fmt");

   586     return FALSE;

   587   }

   588 

   589   wav->format = header->format;

   590   wav->rate = header->rate;

   591   wav->channels = header->channels;

   592   if (wav->channels == 0) {

   593     GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),

   594         ("Stream claims to contain zero channels - invalid data"));

   595     g_free (header);

   596     return FALSE;

   597   }

   598   wav->blockalign = header->blockalign;

   599   wav->width = (header->blockalign * 8) / header->channels;

   600   wav->depth = header->size;

   601   wav->bps = header->av_bps;

   602   if (wav->bps <= 0) {

   603     GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),

   604         ("Stream claims to bitrate of <= zero - invalid data"));

   605     g_free (header);

   606     return FALSE;

   607   }

   608 

   609   /* Note: gst_riff_create_audio_caps might nedd to fix values in

   610    * the header header depending on the format, so call it first */

   611   caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);

   612 

   613   g_free (header);

   614 

   615   if (caps) {

   616     gst_wavparse_create_sourcepad (wav);

   617     gst_pad_use_fixed_caps (wav->srcpad);

   618     gst_pad_set_active (wav->srcpad, TRUE);

   619     gst_pad_set_caps (wav->srcpad, caps);

   620     gst_caps_free (caps);

   621     gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);

   622     gst_element_no_more_pads (GST_ELEMENT (wav));

   623     GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);

   624   } else {

   625     GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));

   626     return FALSE;

   627   }

   628 

   629   return TRUE;

   630 }

   631 

   632 static gboolean

   633 gst_wavparse_other (GstWavParse * wav)

   634 {

   635   guint32 tag, length;

   636 

   637   if (!gst_riff_peek_head (wav, &tag, &length, NULL)) {

   638     GST_WARNING_OBJECT (wav, "could not peek head");

   639     return FALSE;

   640   }

   641   GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,

   642       (gchar *) & tag, length);

   643 

   644   switch (tag) {

   645     case GST_RIFF_TAG_LIST:

   646       if (!(tag = gst_riff_peek_list (wav))) {

   647         GST_WARNING_OBJECT (wav, "could not peek list");

   648         return FALSE;

   649       }

   650 

   651       switch (tag) {

   652         case GST_RIFF_LIST_INFO:

   653           if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) {

   654             GST_WARNING_OBJECT (wav, "could not read list");

   655             return FALSE;

   656           }

   657           break;

   658 

   659         case GST_RIFF_LIST_adtl:

   660           if (!gst_riff_read_skip (wav)) {

   661             GST_WARNING_OBJECT (wav, "could not read skip");

   662             return FALSE;

   663           }

   664           break;

   665 

   666         default:

   667           GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,

   668               (gchar *) & tag);

   669           if (!gst_riff_read_skip (wav)) {

   670             GST_WARNING_OBJECT (wav, "could not read skip");

   671             return FALSE;

   672           }

   673           break;

   674       }

   675 

   676       break;

   677 

   678     case GST_RIFF_TAG_data:

   679       if (!gst_bytestream_flush (wav->bs, 8)) {

   680         GST_WARNING_OBJECT (wav, "could not flush 8 bytes");

   681         return FALSE;

   682       }

   683 

   684       GST_DEBUG_OBJECT (wav, "switching to data mode");

   685       wav->state = GST_WAVPARSE_DATA;

   686       wav->datastart = gst_bytestream_tell (wav->bs);

   687       if (length == 0) {

   688         guint64 file_length;

   689 

   690         /* length is 0, data probably stretches to the end

   691          * of file */

   692         GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");

   693         /* get length of file */

   694         file_length = gst_bytestream_length (wav->bs);

   695         if (file_length == -1) {

   696           GST_DEBUG_OBJECT (wav,

   697               "could not get file length, assuming data to eof");

   698           /* could not get length, assuming till eof */

   699           length = G_MAXUINT32;

   700         }

   701         if (file_length > G_MAXUINT32) {

   702           GST_DEBUG_OBJECT (wav, "file length %lld, clipping to 32 bits");

   703           /* could not get length, assuming till eof */

   704           length = G_MAXUINT32;

   705         } else {

   706           GST_DEBUG_OBJECT (wav, "file length %lld, datalength",

   707               file_length, length);

   708           /* substract offset of datastart from length */

   709           length = file_length - wav->datastart;

   710           GST_DEBUG_OBJECT (wav, "datalength %lld", length);

   711         }

   712       }

   713       wav->datasize = (guint64) length;

   714       break;

   715 

   716     case GST_RIFF_TAG_cue:

   717       if (!gst_riff_read_skip (wav)) {

   718         GST_WARNING_OBJECT (wav, "could not read skip");

   719         return FALSE;

   720       }

   721       break;

   722 

   723     default:

   724       GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);

   725       if (!gst_riff_read_skip (wav))

   726         return FALSE;

   727       break;

   728   }

   729 

   730   return TRUE;

   731 }

   732 #endif

   733 

   734 

   735 

   736 static gboolean

   737 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)

   738 {

   739   guint32 doctype;

   740   

   741   if (!gst_riff_parse_file_header (element, buf, &doctype))

   742    return FALSE;

   743 

   744   if (doctype != GST_RIFF_RIFF_WAVE)

   745     goto not_wav;

   746 

   747   return TRUE;

   748 

   749   /* ERRORS */

   750 not_wav:

   751 

   752   {

   753     GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),

   754         ("File is not an WAVE file: %" GST_FOURCC_FORMAT,

   755             GST_FOURCC_ARGS (doctype)));

   756     return FALSE;

   757   }

   758 }

   759 

   760 static GstFlowReturn

   761 gst_wavparse_stream_init (GstWavParse * wav)

   762 {

   763   GstFlowReturn res;

   764   GstBuffer *buf = NULL;

   765 

   766   

   767   if ((res = gst_pad_pull_range (wav->sinkpad,

   768               wav->offset, 12, &buf)) != GST_FLOW_OK)

   769       {

   770  

   771     return res;

   772       }

   773   else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))

   774     return GST_FLOW_ERROR;

   775 

   776   wav->offset += 12;

   777 

   778   return GST_FLOW_OK;

   779 }

   780 

   781 /* This function is used to perform seeks on the element in 

   782  * pull mode.

   783  *

   784  * It also works when event is NULL, in which case it will just

   785  * start from the last configured segment. This technique is

   786  * used when activating the element and to perform the seek in

   787  * READY.

   788  */

   789 static gboolean

   790 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)

   791 {

   792   gboolean res;

   793   gdouble rate;

   794   GstEvent *newsegment;

   795   GstFormat format;

   796   GstSeekFlags flags;

   797   GstSeekType cur_type, stop_type;

   798   gint64 cur, stop;

   799   gboolean flush;

   800   gboolean update;

   801   GstSegment seeksegment;

   802 

   803   if (event) {

   804     GST_DEBUG_OBJECT (wav, "doing seek with event");

   805 

   806     gst_event_parse_seek (event, &rate, &format, &flags,

   807         &cur_type, &cur, &stop_type, &stop);

   808 

   809     /* we have to have a format as the segment format. Try to convert

   810      * if not. */

   811     if (format != GST_FORMAT_TIME) {

   812       GstFormat fmt;

   813 

   814       fmt = GST_FORMAT_TIME;

   815       res = TRUE;

   816       if (cur_type != GST_SEEK_TYPE_NONE)

   817         res = gst_pad_query_convert (wav->srcpad, format, cur, &fmt, &cur);

   818       if (res && stop_type != GST_SEEK_TYPE_NONE)

   819         res = gst_pad_query_convert (wav->srcpad, format, stop, &fmt, &stop);

   820       if (!res)

   821         goto no_format;

   822 

   823       format = fmt;

   824     }

   825   } else {

   826     GST_DEBUG_OBJECT (wav, "doing seek without event");

   827     flags = 0;

   828   }

   829 

   830   flush = flags & GST_SEEK_FLAG_FLUSH;

   831 

   832   if (flush && wav->srcpad) {

   833     GST_DEBUG_OBJECT (wav, "sending flush start");

   834     gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());

   835   } else {

   836     gst_pad_pause_task (wav->sinkpad);

   837   }

   838 

   839   GST_PAD_STREAM_LOCK (wav->sinkpad);

   840 

   841   /* copy segment, we need this because we still need the old

   842    * segment when we close the current segment. */

   843   memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));

   844 

   845   if (event) {

   846     GST_DEBUG_OBJECT (wav, "configuring seek");

   847     gst_segment_set_seek (&seeksegment, rate, format, flags,

   848         cur_type, cur, stop_type, stop, &update);

   849   }

   850 

   851   if ((stop = seeksegment.stop) == -1)

   852     stop = seeksegment.duration;

   853 

   854   if (cur_type != GST_SEEK_TYPE_NONE) {

   855     wav->offset =

   856         gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND);

   857     wav->offset -= wav->offset % wav->bytes_per_sample;

   858     wav->offset += wav->datastart;

   859   }

   860 

   861   if (stop != -1) {

   862     wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND);

   863     wav->end_offset +=

   864         wav->bytes_per_sample - (wav->end_offset % wav->bytes_per_sample);

   865     wav->end_offset += wav->datastart;

   866   } else {

   867     wav->end_offset = wav->datasize + wav->datastart;

   868   }

   869   wav->offset = MIN (wav->offset, wav->end_offset);

   870   wav->dataleft = wav->end_offset - wav->offset;

   871 

   872   GST_DEBUG_OBJECT (wav,

   873       "seek: offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT ", segment %"

   874       GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, wav->offset, wav->end_offset,

   875       GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));

   876 

   877   /* prepare for streaming again */

   878   if (wav->srcpad) {

   879     if (flush) {

   880       GST_DEBUG_OBJECT (wav, "sending flush stop");

   881       gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());

   882     } else if (wav->segment_running) {

   883       /* we are running the current segment and doing a non-flushing seek,

   884        * close the segment first based on the last_stop. */

   885       GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT

   886           " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);

   887 

   888       gst_pad_push_event (wav->srcpad,

   889           gst_event_new_new_segment (TRUE,

   890               wav->segment.rate, wav->segment.format,

   891               wav->segment.start, wav->segment.last_stop, wav->segment.time));

   892     }

   893   }

   894 

   895   memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));

   896 

   897   if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {

   898     gst_element_post_message (GST_ELEMENT (wav),

   899         gst_message_new_segment_start (GST_OBJECT (wav),

   900             wav->segment.format, wav->segment.last_stop));

   901   }

   902 

   903   /* now send the newsegment */

   904   GST_DEBUG_OBJECT (wav, "Sending newsegment from %" G_GINT64_FORMAT

   905       " to %" G_GINT64_FORMAT, wav->segment.start, stop);

   906 

   907   newsegment =

   908       gst_event_new_new_segment (FALSE, wav->segment.rate,

   909       wav->segment.format, wav->segment.last_stop, stop, wav->segment.time);

   910 

   911   if (wav->srcpad) {

   912     gst_pad_push_event (wav->srcpad, newsegment);

   913   } else {

   914     /* send later when we actually create the source pad */

   915     g_assert (wav->newsegment == NULL);

   916     wav->newsegment = newsegment;

   917   }

   918 

   919   wav->segment_running = TRUE;

   920   if (!wav->streaming) {

   921     gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,

   922         wav->sinkpad);

   923   }

   924 

   925   GST_PAD_STREAM_UNLOCK (wav->sinkpad);

   926 

   927   return TRUE;

   928 

   929   /* ERRORS */

   930 no_format:

   931   {

   932     GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");

   933     return FALSE;

   934   }

   935 }

   936 

   937 /*

   938  * gst_wavparse_peek_chunk_info:

   939  * @wav Wavparse object

   940  * @tag holder for tag

   941  * @size holder for tag size

   942  *                         

   943  * Peek next chunk info (tag and size)                         

   944  *

   945  * Returns: %TRUE when one chunk info has been got from the adapter

   946  */

   947 static gboolean

   948 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)

   949 {

   950   const guint8 *data = NULL;

   951 

   952   if (gst_adapter_available (wav->adapter) < 8) {

   953     return FALSE;

   954   }

   955 

   956   GST_DEBUG ("Next chunk size is %d bytes", *size);

   957   data = gst_adapter_peek (wav->adapter, 8);

   958   *tag = GST_READ_UINT32_LE (data);

   959   *size = GST_READ_UINT32_LE (data + 4);

   960 

   961   return TRUE;

   962 }

   963 

   964 /*

   965  * gst_wavparse_peek_chunk:

   966  * @wav Wavparse object

   967  * @tag holder for tag

   968  * @size holder for tag size

   969  *

   970  * Peek enough data for one full chunk

   971  *

   972  * Returns: %TRUE when one chunk has been got

   973  */

   974 static gboolean

   975 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)

   976 {

   977   guint32 peek_size = 0;

   978 

   979   gst_wavparse_peek_chunk_info (wav, tag, size);

   980   GST_DEBUG ("Need to peek chunk of %d bytes", *size);

   981   peek_size = (*size + 1) & ~1;

   982 

   983   if (gst_adapter_available (wav->adapter) >= (8 + peek_size)) {

   984     return TRUE;

   985   } else {

   986     return FALSE;

   987   }

   988 }

   989 

   990 static gboolean

   991 gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len)

   992 {

   993   gboolean res = FALSE;

   994   GstFormat fmt = GST_FORMAT_BYTES;

   995   GstPad *peer;

   996 

   997   if ((peer = gst_pad_get_peer (wav->sinkpad))) {

   998     res = gst_pad_query_duration (peer, &fmt, len);

   999     gst_object_unref (peer);

  1000   }

  1001 

  1002   return res;

  1003 }

  1004 

  1005 static GstFlowReturn

  1006 gst_wavparse_stream_headers (GstWavParse * wav)

  1007 {

  1008   GstFlowReturn res;

  1009   GstBuffer *buf;

  1010   gst_riff_strf_auds *header = NULL;

  1011   guint32 tag, size;

  1012   gboolean gotdata = FALSE;

  1013   GstCaps *caps;

  1014   gint64 duration;

  1015   gchar *codec_name = NULL;

  1016   GstEvent **event_p;

  1017 

  1018   

  1019   if (!wav->got_fmt) {

  1020     GstBuffer *extra;

  1021 

  1022     /* The header start with a 'fmt ' tag */

  1023 

  1024     if (wav->streaming) {

  1025       if (!gst_wavparse_peek_chunk (wav, &tag, &size))

  1026         return GST_FLOW_OK;

  1027 

  1028       buf = gst_buffer_new ();

  1029       gst_buffer_ref (buf);

  1030       gst_adapter_flush (wav->adapter, 8);

  1031       wav->offset += 8;

  1032       GST_BUFFER_DATA (buf) = (guint8 *) gst_adapter_peek (wav->adapter, size);

  1033       GST_BUFFER_SIZE (buf) = size;

  1034 

  1035     } else {

  1036       if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad,

  1037                   &wav->offset, &tag, &buf)) != GST_FLOW_OK)

  1038         return res;

  1039     }

  1040 

  1041     if (tag != GST_RIFF_TAG_fmt)

  1042       goto invalid_wav;

  1043 

  1044     if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra)))

  1045       goto parse_header_error;

  1046 

  1047     if (wav->streaming) {

  1048       gst_adapter_flush (wav->adapter, size);

  1049       wav->offset += size;

  1050       GST_BUFFER_DATA (buf) = NULL;

  1051       gst_buffer_unref (buf);

  1052     }

  1053 

  1054     /* Note: gst_riff_create_audio_caps might nedd to fix values in

  1055      * the header header depending on the format, so call it first */

  1056     caps =

  1057         gst_riff_create_audio_caps (header->format, NULL, header, extra,

  1058         NULL, &codec_name);

  1059 

  1060     if (extra)

  1061       gst_buffer_unref (extra);

  1062 

  1063     wav->format = header->format;

  1064     wav->rate = header->rate;

  1065     wav->channels = header->channels;

  1066 

  1067     if (wav->channels == 0)

  1068       goto no_channels;

  1069 

  1070     wav->blockalign = header->blockalign;

  1071     wav->width = (header->blockalign * 8) / header->channels;

  1072     wav->depth = header->size;

  1073     wav->bps = header->av_bps;

  1074 

  1075     if (wav->bps <= 0)

  1076       goto no_bitrate;

  1077 

  1078     wav->bytes_per_sample = wav->channels * wav->width / 8;

  1079     if (wav->bytes_per_sample <= 0)

  1080       goto no_bytes_per_sample;

  1081 

  1082     g_free (header);

  1083 

  1084     if (!caps)

  1085       goto unknown_format;

  1086 

  1087     GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);

  1088     GST_DEBUG_OBJECT (wav, "width      = %u", (guint) wav->width);

  1089     GST_DEBUG_OBJECT (wav, "depth      = %u", (guint) wav->depth);

  1090     GST_DEBUG_OBJECT (wav, "bps        = %u", (guint) wav->bps);

  1091 

  1092     /* create pad later so we can sniff the first few bytes

  1093      * of the real data and correct our caps if necessary */

  1094     gst_caps_replace (&wav->caps, caps);

  1095     gst_caps_replace (&caps, NULL);

  1096 

  1097     wav->got_fmt = TRUE;

  1098 

  1099     if (codec_name) {

  1100       wav->tags = gst_tag_list_new ();

  1101 

  1102       gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,

  1103           GST_TAG_AUDIO_CODEC, codec_name, NULL);

  1104 

  1105       g_free (codec_name);

  1106       codec_name = NULL;

  1107     }

  1108 

  1109     GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate,

  1110         wav->channels);

  1111   }

  1112 

  1113   /* loop headers until we get data */

  1114   while (!gotdata) {

  1115     if (wav->streaming) {

  1116       if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))

  1117         return GST_FLOW_OK;

  1118     } else {

  1119       if ((res =

  1120               gst_pad_pull_range (wav->sinkpad, wav->offset, 8,

  1121                   &buf)) != GST_FLOW_OK)

  1122         goto header_read_error;

  1123       tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));

  1124       size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);

  1125     }

  1126 

  1127     /*

  1128        wav is a st00pid format, we don't know for sure where data starts.

  1129        So we have to go bit by bit until we find the 'data' header

  1130      */

  1131 

  1132     switch (tag) {

  1133         /* TODO : Implement the various cases */

  1134       case GST_RIFF_TAG_data:{