gstreamer/gst-libs/gst/audio/audio.h
Sebastian Dröge 846ddaa550 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Use gst_util_uint64_scale() instead of doing the math
with double for GST_FRAMES_TO_CLOCK_TIME() and
GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
prevents rounding errors. Fixes #467667.
2007-08-17 15:24:43 +00:00

154 lines
5 KiB
C

/* 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 */
int gst_audio_frame_byte_size (GstPad* pad);
/* get length in frames of buffer */
long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
GstClockTime gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf);
/* check if the buffer size is a whole multiple of the frame size */
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
*/
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;
#ifndef GST_DISABLE_DEPRECATED
void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);
#endif /* GST_DISABLE_DEPRECATED */
GstBuffer *gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment, gint rate, gint frame_size);
G_END_DECLS
#endif /* __GST_AUDIO_AUDIO_H__ */