gstreamer/gst-libs/gst/audio/audio-converter.h
Tim-Philipp Müller 371e3e460a audio: GST_EXPORT -> GST_AUDIO_API
We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 10:36:56 +00:00

170 lines
6.3 KiB
C

/* GStreamer
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
* (C) 2015 Wim Taymans <wim.taymans@gmail.com>
*
* audioconverter.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_AUDIO_CONVERTER_H__
#define __GST_AUDIO_CONVERTER_H__
#include <gst/gst.h>
#include <gst/audio/audio.h>
G_BEGIN_DECLS
typedef struct _GstAudioConverter GstAudioConverter;
/**
* GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD:
*
* #GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
* changing sample rates.
* Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.
*/
#define GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD "GstAudioConverter.resampler-method"
/**
* GST_AUDIO_CONVERTER_OPT_DITHER_METHOD:
*
* #GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
* changing bit depth.
* Default is #GST_AUDIO_DITHER_NONE.
*/
#define GST_AUDIO_CONVERTER_OPT_DITHER_METHOD "GstAudioConverter.dither-method"
/**
* GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD:
*
* #GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
* to mask noise from quantization errors.
* Default is #GST_AUDIO_NOISE_SHAPING_NONE.
*/
#define GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD "GstAudioConverter.noise-shaping-method"
/**
* GST_AUDIO_CONVERTER_OPT_QUANTIZATION:
*
* #G_TYPE_UINT, The quantization amount. Components will be
* quantized to multiples of this value.
* Default is 1
*/
#define GST_AUDIO_CONVERTER_OPT_QUANTIZATION "GstAudioConverter.quantization"
/**
* GST_AUDIO_CONVERTER_OPT_MIX_MATRIX:
*
* #GST_TYPE_VALUE_LIST, The channel mapping matrix.
*
* The matrix coefficients must be between -1 and 1: the number of rows is equal
* to the number of output channels and the number of columns is equal to the
* number of input channels.
*
* ## Example matrix generation code
* To generate the matrix using code:
*
* |[
* GValue v = G_VALUE_INIT;
* GValue v2 = G_VALUE_INIT;
* GValue v3 = G_VALUE_INIT;
*
* g_value_init (&v2, GST_TYPE_ARRAY);
* g_value_init (&v3, G_TYPE_DOUBLE);
* g_value_set_double (&v3, 1);
* gst_value_array_append_value (&v2, &v3);
* g_value_unset (&v3);
* [ Repeat for as many double as your input channels - unset and reinit v3 ]
* g_value_init (&v, GST_TYPE_ARRAY);
* gst_value_array_append_value (&v, &v2);
* g_value_unset (&v2);
* [ Repeat for as many v2's as your output channels - unset and reinit v2]
* g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v);
* g_value_unset (&v);
* ]|
*/
#define GST_AUDIO_CONVERTER_OPT_MIX_MATRIX "GstAudioConverter.mix-matrix"
/**
* GstAudioConverterFlags:
* @GST_AUDIO_CONVERTER_FLAG_NONE: no flag
* @GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE: the input sample arrays are writable and can be
* used as temporary storage during conversion.
* @GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE: allow arbitrary rate updates with
* gst_audio_converter_update_config().
*
* Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples().
*/
typedef enum {
GST_AUDIO_CONVERTER_FLAG_NONE = 0,
GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE = (1 << 0),
GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE = (1 << 1)
} GstAudioConverterFlags;
GST_AUDIO_API
GstAudioConverter * gst_audio_converter_new (GstAudioConverterFlags flags,
GstAudioInfo *in_info,
GstAudioInfo *out_info,
GstStructure *config);
GST_AUDIO_API
GType gst_audio_converter_get_type (void);
GST_AUDIO_API
void gst_audio_converter_free (GstAudioConverter * convert);
GST_AUDIO_API
void gst_audio_converter_reset (GstAudioConverter * convert);
GST_AUDIO_API
gboolean gst_audio_converter_update_config (GstAudioConverter * convert,
gint in_rate, gint out_rate,
GstStructure *config);
GST_AUDIO_API
const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert,
gint *in_rate, gint *out_rate);
GST_AUDIO_API
gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert,
gsize in_frames);
GST_AUDIO_API
gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert,
gsize out_frames);
GST_AUDIO_API
gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert);
GST_AUDIO_API
gboolean gst_audio_converter_samples (GstAudioConverter * convert,
GstAudioConverterFlags flags,
gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames);
GST_AUDIO_API
gboolean gst_audio_converter_supports_inplace (GstAudioConverter *convert);
GST_AUDIO_API
gboolean gst_audio_converter_convert (GstAudioConverter * convert,
GstAudioConverterFlags flags,
gpointer in, gsize in_size,
gpointer *out, gsize *out_size);
G_END_DECLS
#endif /* __GST_AUDIO_CONVERTER_H__ */