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ed2362149a
Original commit message from CVS: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
119 lines
4.9 KiB
C
119 lines
4.9 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Library <2001> Thomas Vander Stichele <thomas@apestaart.org>
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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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#include <gst/gst.h>
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/* For people that are looking at this source: the purpose of these defines is
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* to make GstCaps a bit easier, in that you don't have to know all of the
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* properties that need to be defined. you can just use these macros. currently
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* (8/01) the only plugins that use these are the passthrough, speed, volume,
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* adder, and [de]interleave plugins.
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* These are for convenience only, and do not specify the 'limits' of
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* GStreamer. you might also use these definitions as a
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* base for making your own caps, if need be.
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*
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* For example, to make a source pad that can output mono streams of either
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* float or int:
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template = gst_pad_template_new
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("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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gst_caps_append(gst_caps_new ("sink_int", "audio/raw",
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GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
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gst_caps_new ("sink_float", "audio/raw",
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GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS)),
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NULL);
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srcpad = gst_pad_new_from_template(template,"src");
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* Andy Wingo, 18 August 2001
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* Thomas, 6 September 2002 */
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/* a few useful defines for arbitrary limits */
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#define GST_AUDIO_MIN_RATE 4000
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#define GST_AUDIO_MAX_RATE 96000
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#define GST_AUDIO_DEF_RATE 44100
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#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \
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gst_props_new (\
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),\
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"signed", GST_PROPS_LIST (\
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GST_PROPS_BOOLEAN (TRUE),\
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GST_PROPS_BOOLEAN (FALSE)\
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),\
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"width", GST_PROPS_LIST (GST_PROPS_INT (8), \
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GST_PROPS_INT (16)), \
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"depth", GST_PROPS_LIST (GST_PROPS_INT (8), \
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GST_PROPS_INT (16)),\
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"rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \
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GST_AUDIO_MAX_RATE),\
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"channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
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NULL)
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#define GST_AUDIO_INT_MONO_PAD_TEMPLATE_PROPS \
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gst_props_new (\
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),\
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"signed", GST_PROPS_LIST (\
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GST_PROPS_BOOLEAN (TRUE),\
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GST_PROPS_BOOLEAN (FALSE)\
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),\
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"width", GST_PROPS_LIST (GST_PROPS_INT (8), \
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GST_PROPS_INT (16)),\
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"depth", GST_PROPS_LIST (GST_PROPS_INT (8), \
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GST_PROPS_INT (16)),\
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"rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \
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GST_AUDIO_MAX_RATE),\
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"channels", GST_PROPS_INT (1),\
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NULL)
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#define GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS \
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gst_props_new (\
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"depth", GST_PROPS_INT (32),\
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),\
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"intercept", GST_PROPS_FLOAT (0.0),\
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"slope", GST_PROPS_FLOAT (1.0),\
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"rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \
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GST_AUDIO_MAX_RATE),\
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"channels", GST_PROPS_INT (1),\
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NULL)
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/*
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* this library defines and implements some helper functions for audio
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* handling
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*/
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/* get byte size of audio frame (based on caps of pad */
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int gst_audio_frame_byte_size (GstPad* pad);
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/* get length in frames of buffer */
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long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
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/* get frame rate based on caps */
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long gst_audio_frame_rate (GstPad *pad);
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/* calculate length in seconds of audio buffer buf based on caps of pad */
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double gst_audio_length (GstPad* pad, GstBuffer* buf);
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/* calculate highest possible sample value based on capabilities of pad */
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long gst_audio_highest_sample_value (GstPad* pad);
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/* check if the buffer size is a whole multiple of the frame size */
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gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
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