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audio lib removed from this dir

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Original commit message from CVS:
audio lib removed from this dir
This commit is contained in:
Thomas Vander Stichele 2001-12-22 23:44:44 +00:00
parent 7ef65f7b85
commit 1fe98d6244
4 changed files with 2 additions and 276 deletions
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6
gst-libs/Makefile.am
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@ -1,5 +1,3 @@
SUBDIRS = audio
# riff getbits putbits idct bytestream control resample
SUBDIRS = gst
DIST_SUBDIRS = audio
# riff getbits putbits idct bytestream control resample
DIST_SUBDIRS = gst

11
gst-libs/audio/Makefile.am
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@ -1,11 +0,0 @@
## libdir = $(libdir)/gst
lib_LTLIBRARIES = libgstaudio.la
libgstaudio_la_SOURCES = gstaudio.c
libgstaudioincludedir = $(includedir)/gst/libs/gstaudio
libgstaudioinclude_HEADERS = gstaudio.h
libgstaudio_la_LIBADD = $(GST_LIBS)
libgstaudio_la_CFLAGS = $(GST_CFLAGS) -finline-functions -ffast-math

152
gst-libs/audio/gstaudio.c
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@ -1,152 +0,0 @@
/* Gnome-Streamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 "gstaudio.h"
int
gst_audio_frame_byte_size (GstPad* pad)
{
/* calculate byte size of an audio frame
* this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
* returns 0 if there's an error, or the byte size if everything's ok
*/
int width = 0;
int channels = 0;
GstCaps *caps = NULL;
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL)
/* ERROR: could not get caps of pad */
return 0;
width = gst_caps_get_int (caps, "width");
channels = gst_caps_get_int (caps, "channels");
return (width / 8) * channels;
}
long
gst_audio_frame_length (GstPad* pad, GstBuffer* buf)
/* calculate length of buffer in frames
* this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
* returns 0 if there's an error, or the number of frames if everything's ok
*/
{
int frame_byte_size = 0;
frame_byte_size = gst_audio_frame_byte_size (pad);
if (frame_byte_size == 0)
/* error */
return 0;
/* FIXME: this function assumes the buffer size to be a whole multiple
* of the frame byte size
*/
return GST_BUFFER_SIZE (buf) / frame_byte_size;
}
long
gst_audio_frame_rate (GstPad *pad)
/*
* calculate frame rate (based on caps of pad)
* returns 0 if failed, rate if success
*/
{
GstCaps *caps = NULL;
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL)
/* ERROR: could not get caps of pad */
return 0;
else
return gst_caps_get_int (caps, "rate");
}
double
gst_audio_length (GstPad* pad, GstBuffer* buf)
{
/* calculate length in seconds
* of audio buffer buf
* based on capabilities of pad
*/
long bytes = 0;
int width = 0;
int channels = 0;
long rate = 0L;
double length;
GstCaps *caps = NULL;
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL)
{
/* ERROR: could not get caps of pad */
length = 0.0;
}
else
{
bytes = GST_BUFFER_SIZE (buf);
width = gst_caps_get_int (caps, "width");
channels = gst_caps_get_int (caps, "channels");
rate = gst_caps_get_int (caps, "rate");
length = (bytes * 8.0) / (double) (rate * channels * width);
}
return length;
}
long
gst_audio_highest_sample_value (GstPad* pad)
/* calculate highest possible sample value
* based on capabilities of pad
*/
{
gboolean is_signed = FALSE;
gint width = 0;
GstCaps *caps = NULL;
caps = GST_PAD_CAPS (pad);
// FIXME : Please change this to a better warning method !
if (caps == NULL)
printf ("WARNING: gstaudio: could not get caps of pad !\n");
width = gst_caps_get_int (caps, "width");
is_signed = gst_caps_get_boolean (caps, "signed");
if (is_signed) --width;
/* example : 16 bit, signed : samples between -32768 and 32767 */
return ((long) (1 << width));
}
gboolean
gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
/* check if the buffer size is a whole multiple of the frame size */
{
if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
return TRUE;
else
return FALSE;
}

109
gst-libs/audio/gstaudio.h
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@ -1,109 +0,0 @@
/* 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);