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41a43b86a8
Original commit message from CVS: port audioconvert to basetransform fix ffmpegcsp and videoscale for basetransform changes
107 lines
3.1 KiB
C
107 lines
3.1 KiB
C
/* GStreamer
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* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
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*
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* gstchannelmix.h: setup of channel conversion matrices
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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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#ifndef __GST_CHANNEL_MIX_H__
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#define __GST_CHANNEL_MIX_H__
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/multichannel.h>
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#define GST_TYPE_AUDIO_CONVERT (gst_audio_convert_get_type())
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#define GST_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
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#define GST_AUDIO_CONVERT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
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#define GST_IS_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CONVERT))
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#define GST_IS_AUDIO_CONVERT_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CONVERT))
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GST_DEBUG_CATEGORY_EXTERN (audio_convert_debug);
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#define GST_CAT_DEFAULT (audio_convert_debug)
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typedef struct _GstAudioConvert GstAudioConvert;
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typedef struct _GstAudioConvertCaps GstAudioConvertCaps;
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typedef struct _GstAudioConvertClass GstAudioConvertClass;
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/* this struct is a handy way of passing around all the caps info ... */
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struct _GstAudioConvertCaps
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{
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/* general caps */
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gboolean is_int;
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gint endianness;
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gint width;
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gint rate;
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gint channels;
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GstAudioChannelPosition *pos;
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/* int audio caps */
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gboolean sign;
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gint depth;
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/* float audio caps */
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gint buffer_frames;
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};
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struct _GstAudioConvert
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{
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GstBaseTransform element;
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GstAudioConvertCaps srccaps;
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GstAudioConvertCaps sinkcaps;
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GstCaps *src_prefered;
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GstCaps *sink_prefered;
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/* channel conversion matrix, m[in_channels][out_channels].
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* If identity matrix, passthrough applies. */
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gfloat **matrix;
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/* conversion functions */
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GstBuffer *(*convert_internal) (GstAudioConvert * this, GstBuffer * buf);
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};
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struct _GstAudioConvertClass
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{
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GstBaseTransformClass parent_class;
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};
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/*
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* Delete channel mixer matrix.
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*/
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void gst_audio_convert_unset_matrix (GstAudioConvert * this);
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/*
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* Setup channel mixer matrix.
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*/
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void gst_audio_convert_setup_matrix (GstAudioConvert * this);
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/*
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* Checks for passthrough (= identity matrix).
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*/
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gboolean gst_audio_convert_passthrough (GstAudioConvert * this);
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/*
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* Do actual mixing.
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*/
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void gst_audio_convert_mix (GstAudioConvert * this,
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gint32 * in_data,
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gint32 * out_data,
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gint samples);
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#endif /* __GST_CHANNEL_MIX_H__ */
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