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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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b957481334
Original commit message from CVS: Fix regressions from using gstriff library
746 lines
21 KiB
C
746 lines
21 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gstwavparse.h>
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static void gst_wavparse_class_init (GstWavParseClass *klass);
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static void gst_wavparse_init (GstWavParse *wavparse);
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static GstElementStateReturn
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gst_wavparse_change_state (GstElement *element);
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static GstCaps* wav_type_find (GstBuffer *buf, gpointer private);
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static const GstFormat* gst_wavparse_get_formats (GstPad *pad);
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static const GstQueryType *
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gst_wavparse_get_query_types (GstPad *pad);
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static gboolean gst_wavparse_pad_query (GstPad *pad,
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GstQueryType type,
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GstFormat *format,
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gint64 *value);
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static gboolean gst_wavparse_pad_convert (GstPad *pad,
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GstFormat src_format,
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gint64 src_value,
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GstFormat *dest_format,
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gint64 *dest_value);
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static void gst_wavparse_chain (GstPad *pad, GstBuffer *buf);
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static const GstEventMask*
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gst_wavparse_get_event_masks (GstPad *pad);
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static gboolean gst_wavparse_srcpad_event (GstPad *pad, GstEvent *event);
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/* elementfactory information */
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static GstElementDetails gst_wavparse_details = {
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".wav parser",
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"Codec/Parser",
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"LGPL",
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"Parse a .wav file into raw audio",
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VERSION,
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"Erik Walthinsen <omega@cse.ogi.edu>",
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"(C) 1999",
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};
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GST_PAD_TEMPLATE_FACTORY (sink_template_factory,
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"wavparse_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"wavparse_wav",
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"audio/x-wav",
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NULL
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)
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)
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GST_PAD_TEMPLATE_FACTORY (src_template_factory,
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"wavparse_src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"wavparse_raw",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_LITTLE_ENDIAN),
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"signed", GST_PROPS_LIST (
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GST_PROPS_BOOLEAN (FALSE),
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GST_PROPS_BOOLEAN (TRUE)
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),
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"width", GST_PROPS_LIST (
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GST_PROPS_INT (8),
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GST_PROPS_INT (16)
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),
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"depth", GST_PROPS_LIST (
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GST_PROPS_INT (8),
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GST_PROPS_INT (16)
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),
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"rate", GST_PROPS_INT_RANGE (8000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2)
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),
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GST_CAPS_NEW (
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"wavparse_mpeg",
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"audio/mpeg",
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"rate", GST_PROPS_INT_RANGE (8000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2),
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"layer", GST_PROPS_INT_RANGE (1, 3)
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),
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GST_CAPS_NEW (
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"parsewav_law",
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"audio/x-alaw",
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"rate", GST_PROPS_INT_RANGE (8000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2)
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),
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GST_CAPS_NEW (
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"parsewav_law",
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"audio/x-mulaw",
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"rate", GST_PROPS_INT_RANGE (8000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2)
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)
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)
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/* typefactory for 'wav' */
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static GstTypeDefinition
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wavdefinition =
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{
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"wavparse_audio/x-wav",
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"audio/x-wav",
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".wav",
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wav_type_find,
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};
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/* WavParse signals and args */
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enum {
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/* FILL ME */
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LAST_SIGNAL
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};
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enum {
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ARG_0,
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/* FILL ME */
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};
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static GstElementClass *parent_class = NULL;
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/*static guint gst_wavparse_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_wavparse_get_type (void)
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{
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static GType wavparse_type = 0;
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if (!wavparse_type) {
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static const GTypeInfo wavparse_info = {
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sizeof(GstWavParseClass), NULL,
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NULL,
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(GClassInitFunc) gst_wavparse_class_init,
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NULL,
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NULL,
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sizeof(GstWavParse),
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0,
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(GInstanceInitFunc) gst_wavparse_init,
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};
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wavparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse", &wavparse_info, 0);
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}
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return wavparse_type;
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}
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static void
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gst_wavparse_class_init (GstWavParseClass *klass)
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{
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GstElementClass *gstelement_class;
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gstelement_class = (GstElementClass*) klass;
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gstelement_class->change_state = gst_wavparse_change_state;
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}
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static void
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gst_wavparse_init (GstWavParse *wavparse)
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{
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/* sink */
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wavparse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_template_factory), "sink");
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gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
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gst_pad_set_formats_function (wavparse->sinkpad, gst_wavparse_get_formats);
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gst_pad_set_convert_function (wavparse->sinkpad, gst_wavparse_pad_convert);
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gst_pad_set_query_type_function (wavparse->sinkpad,
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gst_wavparse_get_query_types);
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gst_pad_set_query_function (wavparse->sinkpad, gst_wavparse_pad_query);
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/* source */
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wavparse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_template_factory), "src");
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gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
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gst_pad_set_formats_function (wavparse->srcpad, gst_wavparse_get_formats);
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gst_pad_set_convert_function (wavparse->srcpad, gst_wavparse_pad_convert);
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gst_pad_set_query_type_function (wavparse->srcpad,
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gst_wavparse_get_query_types);
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gst_pad_set_query_function (wavparse->srcpad, gst_wavparse_pad_query);
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gst_pad_set_event_function (wavparse->srcpad, gst_wavparse_srcpad_event);
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gst_pad_set_event_mask_function (wavparse->srcpad, gst_wavparse_get_event_masks);
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gst_pad_set_chain_function (wavparse->sinkpad, gst_wavparse_chain);
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wavparse->riff = NULL;
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wavparse->state = GST_WAVPARSE_UNKNOWN;
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wavparse->riff = NULL;
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wavparse->riff_nextlikely = 0;
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wavparse->size = 0;
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wavparse->bps = 0;
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wavparse->offset = 0;
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wavparse->need_discont = FALSE;
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}
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static GstCaps*
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wav_type_find (GstBuffer *buf, gpointer private)
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{
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gchar *data = GST_BUFFER_DATA (buf);
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if (GST_BUFFER_SIZE (buf) < 12) return NULL;
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if (strncmp (&data[0], "RIFF", 4)) return NULL;
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if (strncmp (&data[8], "WAVE", 4)) return NULL;
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return gst_caps_new ("wav_type_find", "audio/x-wav", NULL);
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}
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static void wav_new_chunk_callback(GstRiffChunk *chunk, gpointer data)
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{
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GstWavParse *wavparse;
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wavparse = GST_WAVPARSE (data);
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GST_DEBUG("new tag " GST_FOURCC_FORMAT "\n", GST_FOURCC_ARGS(chunk->id));
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if(chunk->id == GST_RIFF_TAG_fmt){
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GstWavParseFormat *format;
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GstCaps *caps = NULL;
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/* we can gather format information now */
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format = (GstWavParseFormat *)((guchar *) GST_BUFFER_DATA (wavparse->buf) + chunk->offset);
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wavparse->bps = GUINT16_FROM_LE(format->wBlockAlign);
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wavparse->rate = GUINT32_FROM_LE(format->dwSamplesPerSec);
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wavparse->channels = GUINT16_FROM_LE(format->wChannels);
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wavparse->width = GUINT16_FROM_LE(format->wBitsPerSample);
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wavparse->format = GINT16_FROM_LE(format->wFormatTag);
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/* set the caps on the src pad */
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/* FIXME: handle all of the other formats as well */
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switch (wavparse->format)
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{
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case GST_RIFF_WAVE_FORMAT_ALAW:
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case GST_RIFF_WAVE_FORMAT_MULAW: {
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gchar *mime = (wavparse->format == GST_RIFF_WAVE_FORMAT_ALAW) ?
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"audio/x-alaw" : "audio/x-mulaw";
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if (!(wavparse->width == 8)) {
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g_warning("Ignoring invalid width %d",
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wavparse->width);
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return;
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}
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caps = GST_CAPS_NEW (
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"parsewav_src",
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mime,
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"rate", GST_PROPS_INT (wavparse->rate),
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"channels", GST_PROPS_INT (wavparse->channels)
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);
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}
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break;
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case GST_RIFF_WAVE_FORMAT_PCM:
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caps = GST_CAPS_NEW (
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"parsewav_src",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_LITTLE_ENDIAN),
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"signed", GST_PROPS_BOOLEAN ((wavparse->width > 8) ? TRUE : FALSE),
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"width", GST_PROPS_INT (wavparse->width),
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"depth", GST_PROPS_INT (wavparse->width),
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"rate", GST_PROPS_INT (wavparse->rate),
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"channels", GST_PROPS_INT (wavparse->channels)
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);
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break;
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case GST_RIFF_WAVE_FORMAT_MPEGL12:
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case GST_RIFF_WAVE_FORMAT_MPEGL3: {
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gint layer = (wavparse->format == GST_RIFF_WAVE_FORMAT_MPEGL12) ? 2 : 3;
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caps = GST_CAPS_NEW (
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"parsewav_src",
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"audio/mpeg",
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"layer", GST_PROPS_INT (layer),
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"rate", GST_PROPS_INT (wavparse->rate),
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"channels", GST_PROPS_INT (wavparse->channels)
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);
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}
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break;
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default:
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gst_element_error (GST_ELEMENT (wavparse), "wavparse: format %d not handled", wavparse->format);
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return;
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}
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if (gst_pad_try_set_caps (wavparse->srcpad, caps) <= 0) {
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gst_element_error (GST_ELEMENT (wavparse), "Could not set caps");
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return;
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}
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GST_DEBUG ("frequency %d, channels %d",
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wavparse->rate, wavparse->channels);
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/* we're now looking for the data chunk */
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wavparse->state = GST_WAVPARSE_CHUNK_DATA;
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}
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}
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static void
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gst_wavparse_chain (GstPad *pad, GstBuffer *buf)
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{
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GstWavParse *wavparse;
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gboolean buffer_riffed = FALSE; /* so we don't parse twice */
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gulong size;
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g_return_if_fail (pad != NULL);
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (buf != NULL);
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g_return_if_fail (GST_BUFFER_DATA (buf) != NULL);
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wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
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GST_DEBUG ("gst_wavparse_chain: got buffer in '%s'",
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gst_object_get_name (GST_OBJECT (wavparse)));
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size = GST_BUFFER_SIZE (buf);
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wavparse->buf = buf;
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/* walk through the states in priority order */
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/* we're in the data region */
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if (wavparse->state == GST_WAVPARSE_DATA) {
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GstFormat format;
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guint64 maxsize;
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/* we can't go beyond the max length */
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maxsize = wavparse->riff_nextlikely - GST_BUFFER_OFFSET (buf);
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if (maxsize == 0) {
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return;
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} else if (maxsize < size) {
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/* if we're expected to see a new chunk in this buffer */
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GstBuffer *newbuf;
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newbuf = gst_buffer_create_sub (buf, 0, maxsize);
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gst_buffer_unref (buf);
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buf = newbuf;
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size = maxsize;
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wavparse->state = GST_WAVPARSE_OTHER;
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/* I suppose we could signal an EOF at this point, but that may be
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premature. We've stopped data flow, that's the main thing. */
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}
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if (GST_PAD_IS_USABLE (wavparse->srcpad)) {
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format = GST_FORMAT_TIME;
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gst_pad_convert (wavparse->srcpad,
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GST_FORMAT_BYTES,
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wavparse->offset,
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&format,
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&GST_BUFFER_TIMESTAMP (buf));
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if (wavparse->need_discont) {
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if (buf && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
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gst_pad_push (wavparse->srcpad,
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GST_BUFFER (gst_event_new_discontinuous (FALSE,
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GST_FORMAT_BYTES, wavparse->offset,
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GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf),
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NULL)));
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} else {
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gst_pad_push (wavparse->srcpad,
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GST_BUFFER (gst_event_new_discontinuous (FALSE,
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GST_FORMAT_BYTES, wavparse->offset,
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NULL)));
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}
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wavparse->need_discont = FALSE;
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}
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gst_pad_push (wavparse->srcpad, buf);
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} else {
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gst_buffer_unref (buf);
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}
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wavparse->offset += size;
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return;
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}
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if (wavparse->state == GST_WAVPARSE_OTHER) {
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GST_DEBUG ("we're in unknown territory here, not passing on");
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gst_buffer_unref(buf);
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return;
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}
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/* here we deal with parsing out the primary state */
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/* these are sequenced such that in the normal case each (RIFF/WAVE,
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fmt, data) will fire in sequence, as they should */
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/* we're in null state now, look for the RIFF header, start parsing */
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if (wavparse->state == GST_WAVPARSE_UNKNOWN) {
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gint retval;
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GST_DEBUG ("GstWavParse: checking for RIFF format");
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/* create a new RIFF parser */
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wavparse->riff = gst_riff_parser_new (wav_new_chunk_callback, wavparse);
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/* give it the current buffer to start parsing */
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retval = gst_riff_parser_next_buffer (wavparse->riff, buf, 0);
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buffer_riffed = TRUE;
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if (retval < 0) {
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GST_DEBUG ("sorry, isn't RIFF");
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gst_buffer_unref(buf);
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return;
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}
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/* this has to be a file of form WAVE for us to deal with it */
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if (wavparse->riff->form != gst_riff_fourcc_to_id ("WAVE")) {
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GST_DEBUG ("sorry, isn't WAVE");
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gst_buffer_unref(buf);
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return;
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}
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/* at this point we're waiting for the 'fmt ' chunk */
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/* careful, the state may have changed in the callback */
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if (wavparse->state == GST_WAVPARSE_UNKNOWN){
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wavparse->state = GST_WAVPARSE_CHUNK_FMT;
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}
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}
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/* we're now looking for the 'fmt ' chunk to get the audio info */
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if (wavparse->state == GST_WAVPARSE_CHUNK_FMT) {
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GST_DEBUG ("GstWavParse: looking for fmt chunk");
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/* there's a good possibility we may not have parsed this buffer */
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if (buffer_riffed == FALSE) {
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gst_riff_parser_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf));
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buffer_riffed = TRUE;
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}
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return;
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}
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/* now we look for the data chunk */
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if (wavparse->state == GST_WAVPARSE_CHUNK_DATA) {
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GstRiffChunk *datachunk;
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GST_DEBUG ("GstWavParse: looking for data chunk");
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/* again, we might need to parse the buffer */
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if (buffer_riffed == FALSE) {
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gst_riff_parser_next_buffer (wavparse->riff, buf, GST_BUFFER_OFFSET (buf));
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buffer_riffed = TRUE;
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}
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datachunk = gst_riff_parser_get_chunk (wavparse->riff, GST_RIFF_TAG_data);
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if (datachunk != NULL) {
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gulong subsize;
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GstBuffer *newbuf;
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GST_DEBUG ("data begins at %ld", datachunk->offset);
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wavparse->datastart = datachunk->offset;
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/* at this point we can ACK that we have data */
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wavparse->state = GST_WAVPARSE_DATA;
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/* now we construct a new buffer for the remainder */
|
|
subsize = size - datachunk->offset;
|
|
GST_DEBUG ("sending last %ld bytes along as audio", subsize);
|
|
|
|
newbuf = gst_buffer_create_sub (buf, datachunk->offset, subsize);
|
|
gst_buffer_unref (buf);
|
|
|
|
GST_BUFFER_TIMESTAMP (newbuf) = 0;
|
|
|
|
if (GST_PAD_IS_USABLE (wavparse->srcpad))
|
|
gst_pad_push (wavparse->srcpad, newbuf);
|
|
else
|
|
gst_buffer_unref (newbuf);
|
|
|
|
wavparse->offset = subsize;
|
|
|
|
/* now we're ready to go, the next buffer should start data */
|
|
wavparse->state = GST_WAVPARSE_DATA;
|
|
|
|
/* however, we may be expecting another chunk at some point */
|
|
wavparse->riff_nextlikely = gst_riff_parser_get_nextlikely (wavparse->riff);
|
|
|
|
return;
|
|
}
|
|
}
|
|
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
/* 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,
|
|
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));
|
|
|
|
bytes_per_sample = wavparse->channels * wavparse->width / 8;
|
|
if (bytes_per_sample == 0) {
|
|
GST_DEBUG ("bytes_per_sample is 0, probably an mp3 - channels %d, width %d\n",
|
|
wavparse->channels, wavparse->width);
|
|
return FALSE;
|
|
}
|
|
byterate = (glong) (bytes_per_sample * wavparse->rate);
|
|
if (byterate == 0) {
|
|
g_warning ("byterate is 0, internal error\n");
|
|
return FALSE;
|
|
}
|
|
GST_DEBUG ("bytes per sample: %d\n", bytes_per_sample);
|
|
|
|
switch (src_format) {
|
|
case GST_FORMAT_BYTES:
|
|
if (*dest_format == GST_FORMAT_DEFAULT)
|
|
*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_DEFAULT:
|
|
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_DEFAULT)
|
|
*dest_value = src_value * wavparse->rate / GST_SECOND;
|
|
else
|
|
return FALSE;
|
|
|
|
*dest_value = *dest_value & ~(bytes_per_sample - 1);
|
|
break;
|
|
default:
|
|
g_warning ("unhandled format for wavparse\n");
|
|
break;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static const GstQueryType *
|
|
gst_wavparse_get_query_types (GstPad *pad)
|
|
{
|
|
static const GstQueryType types[] = {
|
|
GST_QUERY_TOTAL,
|
|
GST_QUERY_POSITION,
|
|
0
|
|
};
|
|
return types;
|
|
}
|
|
|
|
/* handle queries for location and length in requested format */
|
|
static gboolean
|
|
gst_wavparse_pad_query (GstPad *pad, GstQueryType type,
|
|
GstFormat *format, gint64 *value)
|
|
{
|
|
GstFormat peer_format = GST_FORMAT_BYTES;
|
|
gint64 peer_value;
|
|
GstWavParse *wavparse;
|
|
|
|
/* probe sink's peer pad, convert value, and that's it :) */
|
|
/* FIXME: ideally we'd loop over possible formats of peer instead
|
|
* of only using BYTE */
|
|
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
|
|
if (!gst_pad_query (GST_PAD_PEER (wavparse->sinkpad), type,
|
|
&peer_format, &peer_value)) {
|
|
g_warning ("Could not query sink pad's peer\n");
|
|
return FALSE;
|
|
}
|
|
if (!gst_pad_convert (wavparse->sinkpad, peer_format, peer_value,
|
|
format, value)) {
|
|
g_warning ("Could not query sink pad's peer\n");
|
|
return FALSE;
|
|
}
|
|
GST_DEBUG ("pad_query done, value %" G_GINT64_FORMAT "\n", *value);
|
|
return TRUE;
|
|
}
|
|
|
|
static const GstEventMask*
|
|
gst_wavparse_get_event_masks (GstPad *pad)
|
|
{
|
|
static const GstEventMask gst_wavparse_src_event_masks[] = {
|
|
{ GST_EVENT_SEEK, GST_SEEK_METHOD_SET |
|
|
GST_SEEK_FLAG_FLUSH },
|
|
{ 0, }
|
|
};
|
|
return gst_wavparse_src_event_masks;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_srcpad_event (GstPad *pad, GstEvent *event)
|
|
{
|
|
GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
|
|
gboolean res = FALSE;
|
|
|
|
GST_DEBUG ("event %d", GST_EVENT_TYPE (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
gint64 byteoffset;
|
|
GstFormat format;
|
|
|
|
/* we can only seek when in the DATA state */
|
|
if (wavparse->state != GST_WAVPARSE_DATA) {
|
|
return FALSE;
|
|
}
|
|
|
|
format = GST_FORMAT_BYTES;
|
|
|
|
/* bring format to bytes for the peer element,
|
|
* FIXME be smarter here */
|
|
res = gst_pad_convert (pad,
|
|
GST_EVENT_SEEK_FORMAT (event),
|
|
GST_EVENT_SEEK_OFFSET (event),
|
|
&format,
|
|
&byteoffset);
|
|
|
|
if (res) {
|
|
GstEvent *seek;
|
|
|
|
/* seek to byteoffset + header length */
|
|
seek = gst_event_new_seek (
|
|
GST_FORMAT_BYTES |
|
|
(GST_EVENT_SEEK_TYPE (event) & ~GST_SEEK_FORMAT_MASK),
|
|
byteoffset + (GST_EVENT_SEEK_METHOD (event) == GST_SEEK_METHOD_END ? 0 : wavparse->datastart));
|
|
|
|
res = gst_pad_send_event (GST_PAD_PEER (wavparse->sinkpad), seek);
|
|
|
|
if (res) {
|
|
/* ok, seek worked, update our state */
|
|
wavparse->offset = byteoffset;
|
|
wavparse->need_discont = TRUE;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
gst_event_unref (event);
|
|
return res;
|
|
}
|
|
|
|
static GstElementStateReturn
|
|
gst_wavparse_change_state (GstElement *element)
|
|
{
|
|
GstWavParse *wavparse = GST_WAVPARSE (element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_READY_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_PAUSED_TO_PLAYING:
|
|
break;
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
wavparse->riff = NULL;
|
|
wavparse->state = GST_WAVPARSE_UNKNOWN;
|
|
wavparse->riff_nextlikely = 0;
|
|
wavparse->size = 0;
|
|
wavparse->bps = 0;
|
|
wavparse->offset = 0;
|
|
wavparse->need_discont = FALSE;
|
|
break;
|
|
case GST_STATE_READY_TO_NULL:
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
|
|
return GST_STATE_SUCCESS;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GModule *module, GstPlugin *plugin)
|
|
{
|
|
GstElementFactory *factory;
|
|
GstTypeFactory *type;
|
|
|
|
if(!gst_library_load("gstriff")){
|
|
return FALSE;
|
|
}
|
|
|
|
/* 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
|
|
};
|