gstreamer/gst/audioconvert/audioconvert.h
Jan Schmidt d58def621b Add some documentation comments, and some new headers to be scanned.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.
2008-05-22 22:09:16 +00:00

146 lines
4 KiB
C

/* GStreamer
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* audioconvert.h: audio format conversion library
*
* 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 __AUDIO_CONVERT_H__
#define __AUDIO_CONVERT_H__
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
/**
* GstAudioConvertDithering:
* @DITHER_NONE: No dithering
* @DITHER_RPDF: Rectangular dithering
* @DITHER_TPDF: Triangular dithering (default)
* @DITHER_TPDF_HF: High frequency triangular dithering
*
* Set of available dithering methods when converting audio.
*/
typedef enum
{
DITHER_NONE = 0,
DITHER_RPDF,
DITHER_TPDF,
DITHER_TPDF_HF
} GstAudioConvertDithering;
/**
* GstAudioConvertNoiseShaping:
* @NOISE_SHAPING_NONE: No noise shaping (default)
* @NOISE_SHAPING_ERROR_FEEDBACK: Error feedback
* @NOISE_SHAPING_SIMPLE: Simple 2-pole noise shaping
* @NOISE_SHAPING_MEDIUM: Medium 5-pole noise shaping
* @NOISE_SHAPING_HIGH: High 8-pole noise shaping
*
* Set of available noise shaping methods
*/
typedef enum
{
NOISE_SHAPING_NONE = 0,
NOISE_SHAPING_ERROR_FEEDBACK,
NOISE_SHAPING_SIMPLE,
NOISE_SHAPING_MEDIUM,
NOISE_SHAPING_HIGH
} GstAudioConvertNoiseShaping;
typedef struct _AudioConvertCtx AudioConvertCtx;
typedef struct _AudioConvertFmt AudioConvertFmt;
struct _AudioConvertFmt
{
/* general caps */
gboolean is_int;
gint endianness;
gint width;
gint rate;
gint channels;
GstAudioChannelPosition *pos;
gboolean unpositioned_layout;
/* int audio caps */
gboolean sign;
gint depth;
gint unit_size;
};
typedef void (*AudioConvertUnpack) (gpointer src, gpointer dst, gint scale,
gint count);
typedef void (*AudioConvertPack) (gpointer src, gpointer dst, gint scale,
gint count);
typedef void (*AudioConvertMix) (AudioConvertCtx *, gpointer, gpointer, gint);
typedef void (*AudioConvertQuantize) (AudioConvertCtx * ctx, gpointer src,
gpointer dst, gint count);
struct _AudioConvertCtx
{
AudioConvertFmt in;
AudioConvertFmt out;
AudioConvertUnpack unpack;
AudioConvertPack pack;
/* channel conversion matrix, m[in_channels][out_channels].
* If identity matrix, passthrough applies. */
gfloat **matrix;
/* temp storage for channelmix */
gpointer tmp;
gboolean in_default;
gboolean mix_passthrough;
gboolean out_default;
gpointer tmpbuf;
gint tmpbufsize;
gint in_scale;
gint out_scale;
AudioConvertMix channel_mix;
AudioConvertQuantize quantize;
GstAudioConvertDithering dither;
GstAudioConvertNoiseShaping ns;
/* random number generate for dither noise */
GRand *dither_random;
/* last random number generated per channel for hifreq TPDF dither */
gpointer last_random;
/* contains the past quantization errors, error[out_channels][count] */
gdouble *error_buf;
};
gboolean audio_convert_clean_fmt (AudioConvertFmt * fmt);
gboolean audio_convert_prepare_context (AudioConvertCtx * ctx,
AudioConvertFmt * in, AudioConvertFmt * out,
GstAudioConvertDithering dither, GstAudioConvertNoiseShaping ns);
gboolean audio_convert_get_sizes (AudioConvertCtx * ctx, gint samples,
gint * srcsize, gint * dstsize);
gboolean audio_convert_clean_context (AudioConvertCtx * ctx);
gboolean audio_convert_convert (AudioConvertCtx * ctx, gpointer src,
gpointer dst, gint samples, gboolean src_writable);
#endif /* __AUDIO_CONVERT_H__ */