--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/gst-libs/gst/audio/audio.h Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,177 @@
+/* GStreamer
+ * 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>
+
+#ifndef __GST_AUDIO_AUDIO_H__
+#define __GST_AUDIO_AUDIO_H__
+
+G_BEGIN_DECLS
+
+/* 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,
+ * adder, and [de]interleave plugins. 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 streams of either mono
+ * float or any channel int:
+ *
+ * template = gst_pad_template_new
+ * ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+ * gst_caps_append(gst_caps_new ("sink_int", "audio/x-raw-int",
+ * GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
+ * gst_caps_new ("sink_float", "audio/x-raw-float",
+ * GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)),
+ * NULL);
+ *
+ * sinkpad = gst_pad_new_from_template(template, "sink");
+ *
+ * Andy Wingo, 18 August 2001
+ * Thomas, 6 September 2002 */
+
+/* conversion macros */
+/**
+ * GST_FRAMES_TO_CLOCK_TIME:
+ * @frames: sample frames
+ * @rate: sampling rate
+ *
+ * Calculate clocktime from sample @frames and @rate.
+ */
+#define GST_FRAMES_TO_CLOCK_TIME(frames, rate) \
+ ((GstClockTime) gst_util_uint64_scale (frames, GST_SECOND, rate))
+
+/**
+ * GST_CLOCK_TIME_TO_FRAMES:
+ * @clocktime: clock time
+ * @rate: sampling rate
+ *
+ * Calculate frames from @clocktime and sample @rate.
+ */
+#define GST_CLOCK_TIME_TO_FRAMES(clocktime, rate) \
+ gst_util_uint64_scale (clocktime, rate, GST_SECOND)
+
+/**
+ * GST_AUDIO_DEF_RATE:
+ *
+ * Standard sampling rate used in consumer audio.
+ */
+#define GST_AUDIO_DEF_RATE 44100
+
+#define GST_AUDIO_INT_PAD_TEMPLATE_CAPS \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, MAX ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) { 8, 16, 24, 32 }, " \
+ "depth = (int) [ 1, 32 ], " \
+ "signed = (boolean) { true, false }"
+
+/* "standard" int audio is native order, 16 bit stereo. */
+#define GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) 2, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (boolean) true"
+
+#define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \
+ "audio/x-raw-float, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, MAX ], " \
+ "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " \
+ "width = (int) { 32, 64 }"
+
+/* "standard" float audio is native order, 32 bit mono. */
+#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS \
+ "audio/x-raw-float, " \
+ "width = (int) 32, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) 1, " \
+ "endianness = (int) BYTE_ORDER"
+
+/*
+ * this library defines and implements some helper functions for audio
+ * handling
+ */
+
+/* get byte size of audio frame (based on caps of pad */
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+int gst_audio_frame_byte_size (GstPad* pad);
+
+/* get length in frames of buffer */
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+GstClockTime gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf);
+
+/* check if the buffer size is a whole multiple of the frame size */
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
+
+/* functions useful for _getcaps functions */
+/**
+ * GstAudioFieldFlag:
+ *
+ * Do not use anymore.
+ *
+ * Deprecated: use gst_structure_set() directly
+ */
+#ifndef GST_DISABLE_DEPRECATED
+typedef enum {
+ GST_AUDIO_FIELD_RATE = (1 << 0),
+ GST_AUDIO_FIELD_CHANNELS = (1 << 1),
+ GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
+ GST_AUDIO_FIELD_WIDTH = (1 << 3),
+ GST_AUDIO_FIELD_DEPTH = (1 << 4),
+ GST_AUDIO_FIELD_SIGNED = (1 << 5),
+} GstAudioFieldFlag;
+#endif
+
+#ifndef GST_DISABLE_DEPRECATED
+void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);
+#endif /* GST_DISABLE_DEPRECATED */
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+GstBuffer *gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment, gint rate, gint frame_size);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_AUDIO_H__ */