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Diffstat (limited to 'gst-libs/audio/gstaudio.h')
| -rw-r--r-- | gst-libs/audio/gstaudio.h | 109 |
1 files changed, 0 insertions, 109 deletions
diff --git a/gst-libs/audio/gstaudio.h b/gst-libs/audio/gstaudio.h deleted file mode 100644 index 09ef3ec7..00000000 --- a/gst-libs/audio/gstaudio.h +++ /dev/null @@ -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); - - |