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/* 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__ */
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