mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 10:40:34 +00:00
103 lines
4.2 KiB
C
103 lines
4.2 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>
|
|
|
|
typedef struct _GstAudioConverter GstAudioConverter;
|
|
|
|
/**
|
|
* 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"
|
|
|
|
|
|
/**
|
|
* 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;
|
|
|
|
GstAudioConverter * gst_audio_converter_new (GstAudioConverterFlags flags,
|
|
GstAudioInfo *in_info,
|
|
GstAudioInfo *out_info,
|
|
GstStructure *config);
|
|
|
|
void gst_audio_converter_free (GstAudioConverter * convert);
|
|
|
|
void gst_audio_converter_reset (GstAudioConverter * convert);
|
|
|
|
gboolean gst_audio_converter_update_config (GstAudioConverter * convert,
|
|
gint in_rate, gint out_rate,
|
|
GstStructure *config);
|
|
const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert,
|
|
gint *in_rate, gint *out_rate);
|
|
|
|
gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert,
|
|
gsize in_frames);
|
|
gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert,
|
|
gsize out_frames);
|
|
|
|
gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert);
|
|
|
|
|
|
gboolean gst_audio_converter_samples (GstAudioConverter * convert,
|
|
GstAudioConverterFlags flags,
|
|
gpointer in[], gsize in_frames,
|
|
gpointer out[], gsize out_frames);
|
|
|
|
#endif /* __GST_AUDIO_CONVERTER_H__ */
|