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audio library
Original commit message from CVS: audio library
This commit is contained in:
Thomas Vander Stichele
parent
c8199582d7
commit
80d42031b2
3 changed files with 275 additions and 0 deletions
14
gst-libs/audio/Makefile.am
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14
gst-libs/audio/Makefile.am
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@ -0,0 +1,14 @@
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filterdir = $(libdir)/gst
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filter_LTLIBRARIES = libgstaudio.la
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libgstaudio_la_SOURCES = gstaudio.c
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libgstaudioincludedir = $(includedir)/gst/libs/gstaudio
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libgstaudioinclude_HEADERS = gstaudio.h
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libgstaudio_la_LIBADD = $(GST_LIBS)
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libgstaudio_la_CFLAGS = $(GST_CFLAGS) -finline-functions -ffast-math
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# FIXME is this needed?
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## from merge, kept for reference
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## libgstaudio_la_CFLAGS = -O2 $(FOMIT_FRAME_POINTER) -finline-functions -ffast-math $(GLIB_CFLAGS) $(XML_CFLAGS) $(GST_CFLAGS)
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152
gst-libs/audio/gstaudio.c
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152
gst-libs/audio/gstaudio.c
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/* Gnome-Streamer
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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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#include "gstaudio.h"
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int
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gst_audio_frame_byte_size (GstPad* pad)
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{
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/* calculate byte size of an audio frame
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* this should be moved closer to the gstreamer core
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* and be implemented for every mime type IMO
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* returns 0 if there's an error, or the byte size if everything's ok
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*/
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int width = 0;
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int channels = 0;
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GstCaps *caps = NULL;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL)
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/* ERROR: could not get caps of pad */
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return 0;
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width = gst_caps_get_int (caps, "width");
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channels = gst_caps_get_int (caps, "channels");
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return (width / 8) * channels;
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}
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long
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gst_audio_frame_length (GstPad* pad, GstBuffer* buf)
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/* calculate length of buffer in frames
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* this should be moved closer to the gstreamer core
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* and be implemented for every mime type IMO
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* returns 0 if there's an error, or the number of frames if everything's ok
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*/
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{
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int frame_byte_size = 0;
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frame_byte_size = gst_audio_frame_byte_size (pad);
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if (frame_byte_size == 0)
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/* error */
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return 0;
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/* FIXME: this function assumes the buffer size to be a whole multiple
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* of the frame byte size
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*/
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return GST_BUFFER_SIZE (buf) / frame_byte_size;
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}
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long
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gst_audio_frame_rate (GstPad *pad)
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/*
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* calculate frame rate (based on caps of pad)
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* returns 0 if failed, rate if success
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*/
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{
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GstCaps *caps = NULL;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL)
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/* ERROR: could not get caps of pad */
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return 0;
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else
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return gst_caps_get_int (caps, "rate");
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}
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double
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gst_audio_length (GstPad* pad, GstBuffer* buf)
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{
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/* calculate length in seconds
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* of audio buffer buf
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* based on capabilities of pad
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*/
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long bytes = 0;
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int width = 0;
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int channels = 0;
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long rate = 0L;
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double length;
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GstCaps *caps = NULL;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL)
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{
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/* ERROR: could not get caps of pad */
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length = 0.0;
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}
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else
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{
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bytes = GST_BUFFER_SIZE (buf);
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width = gst_caps_get_int (caps, "width");
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channels = gst_caps_get_int (caps, "channels");
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rate = gst_caps_get_int (caps, "rate");
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length = (bytes * 8.0) / (double) (rate * channels * width);
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}
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return length;
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}
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long
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gst_audio_highest_sample_value (GstPad* pad)
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/* calculate highest possible sample value
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* based on capabilities of pad
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*/
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{
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gboolean is_signed = FALSE;
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gint width = 0;
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GstCaps *caps = NULL;
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caps = GST_PAD_CAPS (pad);
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// FIXME : Please change this to a better warning method !
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if (caps == NULL)
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printf ("WARNING: gstaudio: could not get caps of pad !\n");
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width = gst_caps_get_int (caps, "width");
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is_signed = gst_caps_get_boolean (caps, "signed");
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if (is_signed) --width;
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/* example : 16 bit, signed : samples between -32768 and 32767 */
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return ((long) (1 << width));
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}
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gboolean
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gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
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/* check if the buffer size is a whole multiple of the frame size */
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{
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if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
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return TRUE;
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else
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return FALSE;
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}
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109
gst-libs/audio/gstaudio.h
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109
gst-libs/audio/gstaudio.h
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/* Gnome-Streamer
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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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* and [de]interleave plugins. so. these are for convenience only, and do not
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* specify the 'limits' of 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_padtemplate_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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#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \
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gst_props_new (\
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"format", GST_PROPS_STRING ("int"),\
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"law", GST_PROPS_INT (0),\
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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), GST_PROPS_INT(16)),\
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"depth", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\
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"rate", GST_PROPS_INT_RANGE (4000, 96000),\
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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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"format", GST_PROPS_STRING ("int"),\
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"law", GST_PROPS_INT (0),\
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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), GST_PROPS_INT(16)),\
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"depth", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\
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"rate", GST_PROPS_INT_RANGE (4000, 96000),\
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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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"format", GST_PROPS_STRING ("float"),\
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"layout", GST_PROPS_STRING ("gfloat"),\
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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 (4000, 96000),\
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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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