mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 13:41:48 +00:00
109 lines
4.6 KiB
C
109 lines
4.6 KiB
C
/* Gnome-Streamer
|
|
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
|
|
* Library <2001> Thomas Vander Stichele <thomas@apestaart.org>
|
|
*
|
|
* 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 <gst/gst.h>
|
|
|
|
/* for people that are looking at this source: the purpose of these defines is
|
|
* to make GstCaps a bit easier, in that you don't have to know all of the
|
|
* properties that need to be defined. you can just use these macros. currently
|
|
* (8/01) the only plugins that use these are the passthrough, speed, volume,
|
|
* and [de]interleave plugins. so. these are for convenience only, and do not
|
|
* specify the 'limits' of gstreamer. you might also use these definitions as a
|
|
* base for making your own caps, if need be.
|
|
*
|
|
* for example, to make a source pad that can output mono streams of either
|
|
* float or int:
|
|
|
|
template = gst_padtemplate_new
|
|
("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
|
|
gst_caps_append(gst_caps_new ("sink_int", "audio/raw",
|
|
GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
|
|
gst_caps_new ("sink_float", "audio/raw",
|
|
GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS)),
|
|
NULL);
|
|
|
|
srcpad = gst_pad_new_from_template(template,"src");
|
|
|
|
* Andy Wingo, 18 August 2001 */
|
|
|
|
#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \
|
|
gst_props_new (\
|
|
"format", GST_PROPS_STRING ("int"),\
|
|
"law", GST_PROPS_INT (0),\
|
|
"endianness", GST_PROPS_INT (G_BYTE_ORDER),\
|
|
"signed", GST_PROPS_LIST (\
|
|
GST_PROPS_BOOLEAN (TRUE),\
|
|
GST_PROPS_BOOLEAN(FALSE)\
|
|
),\
|
|
"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 (4000, 96000),\
|
|
"channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
|
|
NULL)
|
|
|
|
#define GST_AUDIO_INT_MONO_PAD_TEMPLATE_PROPS \
|
|
gst_props_new (\
|
|
"format", GST_PROPS_STRING ("int"),\
|
|
"law", GST_PROPS_INT (0),\
|
|
"endianness", GST_PROPS_INT (G_BYTE_ORDER),\
|
|
"signed", GST_PROPS_LIST (\
|
|
GST_PROPS_BOOLEAN (TRUE),\
|
|
GST_PROPS_BOOLEAN(FALSE)\
|
|
),\
|
|
"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 (4000, 96000),\
|
|
"channels", GST_PROPS_INT (1),\
|
|
NULL)
|
|
|
|
#define GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS \
|
|
gst_props_new (\
|
|
"format", GST_PROPS_STRING ("float"),\
|
|
"layout", GST_PROPS_STRING ("gfloat"),\
|
|
"intercept", GST_PROPS_FLOAT (0.0),\
|
|
"slope", GST_PROPS_FLOAT (1.0),\
|
|
"rate", GST_PROPS_INT_RANGE (4000, 96000),\
|
|
"channels", GST_PROPS_INT (1),\
|
|
NULL)
|
|
|
|
/*
|
|
* this library defines and implements some helper functions for audio
|
|
* handling
|
|
*/
|
|
|
|
/* get byte size of audio frame (based on caps of pad */
|
|
int gst_audio_frame_byte_size (GstPad* pad);
|
|
|
|
/* get length in frames of buffer */
|
|
long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
|
|
|
|
/* get frame rate based on caps */
|
|
long gst_audio_frame_rate (GstPad *pad);
|
|
|
|
/* calculate length in seconds of audio buffer buf based on caps of pad */
|
|
double gst_audio_length (GstPad* pad, GstBuffer* buf);
|
|
|
|
/* calculate highest possible sample value based on capabilities of pad */
|
|
long gst_audio_highest_sample_value (GstPad* pad);
|
|
|
|
/* check if the buffer size is a whole multiple of the frame size */
|
|
gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
|
|
|
|
|