/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #include #include static void gst_wavparse_class_init (GstWavParseClass *klass); static void gst_wavparse_init (GstWavParse *wavparse); static GstCaps* wav_type_find (GstBuffer *buf, gpointer private); static const GstFormat* gst_wavparse_get_formats (GstPad *pad); static const GstQueryType * gst_wavparse_get_query_types (GstPad *pad); static gboolean gst_wavparse_pad_query (GstPad *pad, GstQueryType type, GstFormat *format, gint64 *value); static gboolean gst_wavparse_pad_convert (GstPad *pad, GstFormat src_format, gint64 src_value, GstFormat *dest_format, gint64 *dest_value); static void gst_wavparse_chain (GstPad *pad, GstBuffer *buf); /* elementfactory information */ static GstElementDetails gst_wavparse_details = { ".wav parser", "Codec/Parser", "LGPL", "Parse a .wav file into raw audio", VERSION, "Erik Walthinsen ", "(C) 1999", }; GST_PAD_TEMPLATE_FACTORY (sink_template_factory, "wavparse_sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_CAPS_NEW ( "wavparse_wav", "audio/x-wav", NULL ) ) GST_PAD_TEMPLATE_FACTORY (src_template_factory, "wavparse_src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_CAPS_NEW ( "wavparse_raw", "audio/raw", "format", GST_PROPS_STRING ("int"), "law", GST_PROPS_INT (0), "endianness", GST_PROPS_INT (G_BYTE_ORDER), "signed", GST_PROPS_LIST ( GST_PROPS_BOOLEAN (FALSE), GST_PROPS_BOOLEAN (TRUE) ), "width", GST_PROPS_LIST ( GST_PROPS_INT (8), GST_PROPS_INT (16) ), "depth", GST_PROPS_LIST ( GST_PROPS_INT (8), GST_PROPS_INT (16) ), "rate", GST_PROPS_INT_RANGE (8000, 48000), "channels", GST_PROPS_INT_RANGE (1, 2) ), GST_CAPS_NEW ( "wavparse_mp3", "audio/x-mp3", NULL ) ) /* typefactory for 'wav' */ static GstTypeDefinition wavdefinition = { "wavparse_audio/x-wav", "audio/x-wav", ".wav", wav_type_find, }; /* WavParse signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0, /* FILL ME */ }; static GstElementClass *parent_class = NULL; /*static guint gst_wavparse_signals[LAST_SIGNAL] = { 0 }; */ GType gst_wavparse_get_type (void) { static GType wavparse_type = 0; if (!wavparse_type) { static const GTypeInfo wavparse_info = { sizeof(GstWavParseClass), NULL, NULL, (GClassInitFunc) gst_wavparse_class_init, NULL, NULL, sizeof(GstWavParse), 0, (GInstanceInitFunc) gst_wavparse_init, }; wavparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse", &wavparse_info, 0); } return wavparse_type; } static void gst_wavparse_class_init (GstWavParseClass *klass) { GstElementClass *gstelement_class; gstelement_class = (GstElementClass*) klass; parent_class = g_type_class_ref (GST_TYPE_ELEMENT); } static void gst_wavparse_init (GstWavParse *wavparse) { /* sink */ wavparse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_template_factory), "sink"); gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad); gst_pad_set_formats_function (wavparse->sinkpad, gst_wavparse_get_formats); gst_pad_set_convert_function (wavparse->sinkpad, gst_wavparse_pad_convert); gst_pad_set_query_type_function (wavparse->sinkpad, gst_wavparse_get_query_types); gst_pad_set_query_function (wavparse->sinkpad, gst_wavparse_pad_query); /* source */ wavparse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_template_factory), "src"); gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->srcpad); gst_pad_set_formats_function (wavparse->srcpad, gst_wavparse_get_formats); gst_pad_set_convert_function (wavparse->srcpad, gst_wavparse_pad_convert); gst_pad_set_query_type_function (wavparse->srcpad, gst_wavparse_get_query_types); gst_pad_set_query_function (wavparse->srcpad, gst_wavparse_pad_query); gst_pad_set_chain_function (wavparse->sinkpad, gst_wavparse_chain); wavparse->riff = NULL; wavparse->state = GST_WAVPARSE_UNKNOWN; wavparse->riff = NULL; wavparse->riff_nextlikely = 0; wavparse->size = 0; wavparse->bps = 0; } static GstCaps* wav_type_find (GstBuffer *buf, gpointer private) { gchar *data = GST_BUFFER_DATA (buf); if (strncmp (&data[0], "RIFF", 4)) return NULL; if (strncmp (&data[8], "WAVE", 4)) return NULL; return gst_caps_new ("wav_type_find", "audio/x-wav", NULL); } /* set timestamp on outgoing buffer * returns TRUE if a timestamp was set */ static gboolean gst_wavparse_set_timestamp (GstWavParse *wavparse, GstBuffer *buf) { gboolean retval = FALSE; /* only do timestamps on linear audio */ switch (wavparse->format) { case GST_RIFF_WAVE_FORMAT_PCM: GST_BUFFER_TIMESTAMP (buf) = wavparse->offset * GST_SECOND / wavparse->rate; wavparse->offset += GST_BUFFER_SIZE (buf) * 8 / (wavparse->width * wavparse->channels); retval = TRUE; break; default: break; } return retval; } static void gst_wavparse_chain (GstPad *pad, GstBuffer *buf) { GstWavParse *wavparse; gboolean buffer_riffed = FALSE; /* so we don't parse twice */ gchar *data; gulong size; g_return_if_fail (pad != NULL); g_return_if_fail (GST_IS_PAD (pad)); g_return_if_fail (buf != NULL); g_return_if_fail (GST_BUFFER_DATA (buf) != NULL); wavparse = GST_WAVPARSE (gst_pad_get_parent (pad)); GST_DEBUG (0, "gst_wavparse_chain: got buffer in '%s'", gst_object_get_name (GST_OBJECT (wavparse))); data = (guchar *) GST_BUFFER_DATA (buf); size = GST_BUFFER_SIZE (buf); /* walk through the states in priority order */ /* we're in the data region */ if (wavparse->state == GST_WAVPARSE_DATA) { /* if we're expected to see a new chunk in this buffer */ if ((wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf)) < GST_BUFFER_SIZE (buf)) { GST_BUFFER_SIZE (buf) = wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf); wavparse->state = GST_WAVPARSE_OTHER; /* I suppose we could signal an EOF at this point, but that may be premature. We've stopped data flow, that's the main thing. */ } gst_wavparse_set_timestamp (wavparse, buf); if (GST_PAD_IS_USABLE (wavparse->srcpad)) gst_pad_push (wavparse->srcpad, buf); else gst_buffer_unref (buf); return; } if (wavparse->state == GST_WAVPARSE_OTHER) { GST_DEBUG (0, "we're in unknown territory here, not passing on"); return; } /* here we deal with parsing out the primary state */ /* these are sequenced such that in the normal case each (RIFF/WAVE, fmt, data) will fire in sequence, as they should */ /* we're in null state now, look for the RIFF header, start parsing */ if (wavparse->state == GST_WAVPARSE_UNKNOWN) { gint retval; GST_DEBUG (0, "GstWavParse: checking for RIFF format"); /* create a new RIFF parser */ wavparse->riff = gst_riff_new (); /* give it the current buffer to start parsing */ retval = gst_riff_next_buffer (wavparse->riff, buf, 0); buffer_riffed = TRUE; if (retval < 0) { GST_DEBUG (0, "sorry, isn't RIFF"); return; } /* this has to be a file of form WAVE for us to deal with it */ if (wavparse->riff->form != gst_riff_fourcc_to_id ("WAVE")) { GST_DEBUG (0, "sorry, isn't WAVE"); return; } /* at this point we're waiting for the 'fmt ' chunk */ wavparse->state = GST_WAVPARSE_CHUNK_FMT; } /* we're now looking for the 'fmt ' chunk to get the audio info */ if (wavparse->state == GST_WAVPARSE_CHUNK_FMT) { GstRiffChunk *fmt; GstWavParseFormat *format; GST_DEBUG (0, "GstWavParse: looking for fmt chunk"); /* there's a good possibility we may not have parsed this buffer */ if (buffer_riffed == FALSE) { gst_riff_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf)); buffer_riffed = TRUE; } /* see if the fmt chunk is available yet */ fmt = gst_riff_get_chunk (wavparse->riff, "fmt "); /* if we've got something, deal with it */ if (fmt != NULL) { GstCaps *caps = NULL; /* we can gather format information now */ format = (GstWavParseFormat *)((guchar *) GST_BUFFER_DATA (buf) + fmt->offset); wavparse->bps = GUINT16_FROM_LE(format->wBlockAlign); wavparse->rate = GUINT32_FROM_LE(format->dwSamplesPerSec); wavparse->channels = GUINT16_FROM_LE(format->wChannels); wavparse->width = GUINT16_FROM_LE(format->wBitsPerSample); wavparse->format = GINT16_FROM_LE(format->wFormatTag); /* set the caps on the src pad */ /* FIXME: handle all of the other formats as well */ switch (wavparse->format) { case GST_RIFF_WAVE_FORMAT_PCM: caps = GST_CAPS_NEW ( "parsewav_src", "audio/raw", "format", GST_PROPS_STRING ("int"), "law", GST_PROPS_INT (0), /*FIXME */ "endianness", GST_PROPS_INT (G_BYTE_ORDER), "signed", GST_PROPS_BOOLEAN ((wavparse->width > 8) ? TRUE : FALSE), "width", GST_PROPS_INT (wavparse->width), "depth", GST_PROPS_INT (wavparse->width), "rate", GST_PROPS_INT (wavparse->rate), "channels", GST_PROPS_INT (wavparse->channels) ); break; case GST_RIFF_WAVE_FORMAT_MPEGL12: case GST_RIFF_WAVE_FORMAT_MPEGL3: caps = GST_CAPS_NEW ( "parsewav_src", "audio/x-mp3", NULL ); break; default: g_warning ("wavparse: format %d not handled", wavparse->format); } if (gst_pad_try_set_caps (wavparse->srcpad, caps) <= 0) { gst_element_error (GST_ELEMENT (wavparse), "Could not set caps"); return; } GST_DEBUG (0, "frequency %d, channels %d", wavparse->rate, wavparse->channels); /* we're now looking for the data chunk */ wavparse->state = GST_WAVPARSE_CHUNK_DATA; } else { /* otherwise we just sort of give up for this buffer */ gst_buffer_unref (buf); return; } } /* now we look for the data chunk */ if (wavparse->state == GST_WAVPARSE_CHUNK_DATA) { GstBuffer *newbuf; GstRiffChunk *datachunk; GST_DEBUG (0, "GstWavParse: looking for data chunk"); /* again, we might need to parse the buffer */ if (buffer_riffed == FALSE) { gst_riff_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf)); buffer_riffed = TRUE; } datachunk = gst_riff_get_chunk (wavparse->riff, "data"); if (datachunk != NULL) { gulong subsize; GST_DEBUG (0, "data begins at %ld", datachunk->offset); /* at this point we can ACK that we have data */ wavparse->state = GST_WAVPARSE_DATA; /* now we construct a new buffer for the remainder */ subsize = size - datachunk->offset; GST_DEBUG (0, "sending last %ld bytes along as audio", subsize); newbuf = gst_buffer_new (); GST_BUFFER_DATA (newbuf) = g_malloc (subsize); GST_BUFFER_SIZE (newbuf) = subsize; gst_wavparse_set_timestamp (wavparse, newbuf); memcpy (GST_BUFFER_DATA (newbuf), GST_BUFFER_DATA (buf) + datachunk->offset, subsize); gst_buffer_unref (buf); if (GST_PAD_IS_USABLE (wavparse->srcpad)) gst_pad_push (wavparse->srcpad, newbuf); else gst_buffer_unref (newbuf); /* now we're ready to go, the next buffer should start data */ wavparse->state = GST_WAVPARSE_DATA; /* however, we may be expecting another chunk at some point */ wavparse->riff_nextlikely = gst_riff_get_nextlikely (wavparse->riff); } else { /* otherwise we just sort of give up for this buffer */ gst_buffer_unref (buf); return; } } } /* convert and query stuff */ static const GstFormat * gst_wavparse_get_formats (GstPad *pad) { static GstFormat formats[] = { GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_UNITS, /* a "frame", ie a set of samples per Hz */ 0, 0 }; return formats; } static gboolean gst_wavparse_pad_convert (GstPad *pad, GstFormat src_format, gint64 src_value, GstFormat *dest_format, gint64 *dest_value) { gint bytes_per_sample; glong byterate; GstWavParse *wavparse; wavparse = GST_WAVPARSE (gst_pad_get_parent (pad)); if (*dest_format == GST_FORMAT_DEFAULT) *dest_format = GST_FORMAT_TIME; bytes_per_sample = wavparse->channels * wavparse->width / 8; if (bytes_per_sample == 0) { g_warning ("bytes_per_sample is 0, internal error\n"); g_warning ("channels %d, width %d\n", wavparse->channels, wavparse->width); return FALSE; } byterate = (glong) (bytes_per_sample * wavparse->rate); if (byterate == 0) { g_warning ("byterate is 0, internal error\n"); return FALSE; } g_print ("DEBUG: bytes per sample: %d\n", bytes_per_sample); switch (src_format) { case GST_FORMAT_BYTES: if (*dest_format == GST_FORMAT_UNITS) *dest_value = src_value / bytes_per_sample; else if (*dest_format == GST_FORMAT_TIME) *dest_value = src_value * GST_SECOND / byterate; else return FALSE; break; case GST_FORMAT_UNITS: if (*dest_format == GST_FORMAT_BYTES) *dest_value = src_value * bytes_per_sample; else if (*dest_format == GST_FORMAT_TIME) *dest_value = src_value * GST_SECOND / wavparse->rate; else return FALSE; break; case GST_FORMAT_TIME: if (*dest_format == GST_FORMAT_BYTES) *dest_value = src_value * byterate / GST_SECOND; else if (*dest_format == GST_FORMAT_UNITS) *dest_value = src_value * wavparse->rate /GST_SECOND; else return FALSE; break; default: g_warning ("unhandled format for wavparse\n"); break; } return TRUE; } static const GstQueryType * gst_wavparse_get_query_types (GstPad *pad) { static const GstQueryType types[] = { GST_QUERY_TOTAL, GST_QUERY_POSITION, 0 }; return types; } /* handle queries for location and length in requested format */ static gboolean gst_wavparse_pad_query (GstPad *pad, GstQueryType type, GstFormat *format, gint64 *value) { GstFormat peer_format = GST_FORMAT_BYTES; gint64 peer_value; GstWavParse *wavparse; /* probe sink's peer pad, convert value, and that's it :) */ /* FIXME: ideally we'd loop over possible formats of peer instead * of only using BYTE */ wavparse = GST_WAVPARSE (gst_pad_get_parent (pad)); if (!gst_pad_query (GST_PAD_PEER (wavparse->sinkpad), type, &peer_format, &peer_value)) { g_warning ("Could not query sink pad's peer\n"); return FALSE; } if (!gst_pad_convert (wavparse->sinkpad, peer_format, peer_value, format, value)) { g_warning ("Could not query sink pad's peer\n"); return FALSE; } g_print ("DEBUG: pad_query done, value %" G_GINT64_FORMAT "\n", *value); return TRUE; } static gboolean plugin_init (GModule *module, GstPlugin *plugin) { GstElementFactory *factory; GstTypeFactory *type; /* create an elementfactory for the wavparse element */ factory = gst_element_factory_new ("wavparse", GST_TYPE_WAVPARSE, &gst_wavparse_details); g_return_val_if_fail(factory != NULL, FALSE); gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_SECONDARY); /* register src pads */ gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (sink_template_factory)); gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (src_template_factory)); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); type = gst_type_factory_new (&wavdefinition); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type)); return TRUE; } GstPluginDesc plugin_desc = { GST_VERSION_MAJOR, GST_VERSION_MINOR, "wavparse", plugin_init };