mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 18:50:48 +00:00
1dae961cbf
Original commit message from CVS: Plugin port to 0.9, ogg/theora playback should work in the seek example now. Removed old examples. Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as explained in 0.9 TODO doc.
110 lines
3.1 KiB
C
110 lines
3.1 KiB
C
/* GStreamer
|
|
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
|
|
*
|
|
* gstchannelmix.h: setup of channel conversion matrices
|
|
*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef __GST_CHANNEL_MIX_H__
|
|
#define __GST_CHANNEL_MIX_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/audio/multichannel.h>
|
|
|
|
#define GST_TYPE_AUDIO_CONVERT (gst_audio_convert_get_type())
|
|
#define GST_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
|
|
#define GST_AUDIO_CONVERT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
|
|
#define GST_IS_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CONVERT))
|
|
#define GST_IS_AUDIO_CONVERT_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CONVERT))
|
|
|
|
GST_DEBUG_CATEGORY_EXTERN (audio_convert_debug);
|
|
#define GST_CAT_DEFAULT (audio_convert_debug)
|
|
|
|
typedef struct _GstAudioConvert GstAudioConvert;
|
|
typedef struct _GstAudioConvertCaps GstAudioConvertCaps;
|
|
typedef struct _GstAudioConvertClass GstAudioConvertClass;
|
|
|
|
/* this struct is a handy way of passing around all the caps info ... */
|
|
struct _GstAudioConvertCaps
|
|
{
|
|
/* general caps */
|
|
gboolean is_int;
|
|
gint endianness;
|
|
gint width;
|
|
gint rate;
|
|
gint channels;
|
|
GstAudioChannelPosition *pos;
|
|
|
|
/* int audio caps */
|
|
gboolean sign;
|
|
gint depth;
|
|
|
|
/* float audio caps */
|
|
gint buffer_frames;
|
|
};
|
|
|
|
struct _GstAudioConvert
|
|
{
|
|
GstElement element;
|
|
|
|
/* pads */
|
|
GstPad *sink;
|
|
GstPad *src;
|
|
|
|
GstAudioConvertCaps srccaps;
|
|
GstAudioConvertCaps sinkcaps;
|
|
|
|
GstCaps *src_prefered;
|
|
GstCaps *sink_prefered;
|
|
|
|
/* channel conversion matrix, m[in_channels][out_channels].
|
|
* If identity matrix, passthrough applies. */
|
|
gfloat **matrix;
|
|
|
|
/* conversion functions */
|
|
GstBuffer *(*convert_internal) (GstAudioConvert * this, GstBuffer * buf);
|
|
};
|
|
|
|
struct _GstAudioConvertClass
|
|
{
|
|
GstElementClass parent_class;
|
|
};
|
|
|
|
/*
|
|
* Delete channel mixer matrix.
|
|
*/
|
|
void gst_audio_convert_unset_matrix (GstAudioConvert * this);
|
|
|
|
/*
|
|
* Setup channel mixer matrix.
|
|
*/
|
|
void gst_audio_convert_setup_matrix (GstAudioConvert * this);
|
|
|
|
/*
|
|
* Checks for passthrough (= identity matrix).
|
|
*/
|
|
gboolean gst_audio_convert_passthrough (GstAudioConvert * this);
|
|
|
|
/*
|
|
* Do actual mixing.
|
|
*/
|
|
void gst_audio_convert_mix (GstAudioConvert * this,
|
|
gint32 * in_data,
|
|
gint32 * out_data,
|
|
gint samples);
|
|
|
|
#endif /* __GST_CHANNEL_MIX_H__ */
|