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6be2524031
Original commit message from CVS: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip): * gst-libs/gst/audio/audio.h: * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite): API: Add buffer clipping function for raw audio buffers. Fixes #456656. Also add deprecation guards for gst_audio_structure_set_int() to the header.
154 lines
5.1 KiB
C
154 lines
5.1 KiB
C
/* GStreamer
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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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#ifndef __GST_AUDIO_AUDIO_H__
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#define __GST_AUDIO_AUDIO_H__
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G_BEGIN_DECLS
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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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* adder, and [de]interleave plugins. 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 streams of either mono
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* float or any channel int:
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*
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* template = gst_pad_template_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/x-raw-int",
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* GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
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* gst_caps_new ("sink_float", "audio/x-raw-float",
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* GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)),
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* NULL);
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*
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* sinkpad = gst_pad_new_from_template(template, "sink");
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*
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* Andy Wingo, 18 August 2001
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* Thomas, 6 September 2002 */
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/* conversion macros */
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/**
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* GST_FRAMES_TO_CLOCK_TIME:
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* @frames: sample frames
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* @rate: sampling rate
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*
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* Calculate clocktime from sample @frames and @rate.
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*/
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#define GST_FRAMES_TO_CLOCK_TIME(frames, rate) \
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((GstClockTime) (((gdouble) frames / rate) * GST_SECOND))
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/**
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* GST_CLOCK_TIME_TO_FRAMES:
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* @clocktime: clock time
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* @rate: sampling rate
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*
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* Calculate frames from @clocktime and sample @rate.
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*/
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#define GST_CLOCK_TIME_TO_FRAMES(clocktime, rate) \
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((gint64) ((gst_guint64_to_gdouble (clocktime) / GST_SECOND) * rate))
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/**
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* GST_AUDIO_DEF_RATE:
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*
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* Standard sampling rate used in consumer audio.
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*/
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#define GST_AUDIO_DEF_RATE 44100
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#define GST_AUDIO_INT_PAD_TEMPLATE_CAPS \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) { 8, 16, 24, 32 }, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }"
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/* "standard" int audio is native order, 16 bit stereo. */
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#define GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) 2, " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"signed = (boolean) true"
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#define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " \
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"width = (int) { 32, 64 }"
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/* "standard" float audio is native order, 32 bit mono. */
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#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS \
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"audio/x-raw-float, " \
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"width = (int) 32, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) 1, " \
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"endianness = (int) BYTE_ORDER"
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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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GstClockTime gst_audio_duration_from_pad_buffer (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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gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
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/* functions useful for _getcaps functions */
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/**
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* GstAudioFieldFlag:
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*
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* Do not use anymore.
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* @Deprecated: use gst_structure_set() directly
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*/
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typedef enum {
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GST_AUDIO_FIELD_RATE = (1 << 0),
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GST_AUDIO_FIELD_CHANNELS = (1 << 1),
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GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
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GST_AUDIO_FIELD_WIDTH = (1 << 3),
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GST_AUDIO_FIELD_DEPTH = (1 << 4),
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GST_AUDIO_FIELD_SIGNED = (1 << 5),
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} GstAudioFieldFlag;
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#ifndef GST_DISABLE_DEPRECATED
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void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);
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#endif /* GST_DISABLE_DEPRECATED */
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GstBuffer *gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment, gint rate, gint frame_size);
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G_END_DECLS
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#endif /* __GST_AUDIO_AUDIO_H__ */
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