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authorThomas Vander Stichele <thomas@apestaart.org>2001-12-21 10:12:38 +0000
committerThomas Vander Stichele <thomas@apestaart.org>2001-12-21 10:12:38 +0000
commit2504f9e1b4ba28ab02f5f7c238952e62cbdebd15 (patch)
treec9f67e93578515f82a395de0c3d36a6796b33eab /gst-libs/audio/gstaudio.h
parent99b237bf25dfe13d4d398832c114174f8355d8ac (diff)
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audio library
Original commit message from CVS: audio library
Diffstat (limited to 'gst-libs/audio/gstaudio.h')
-rw-r--r--gst-libs/audio/gstaudio.h109
1 files changed, 109 insertions, 0 deletions
diff --git a/gst-libs/audio/gstaudio.h b/gst-libs/audio/gstaudio.h
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+/* 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);
+
+