/* GStreamer
* Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
*
* 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_RESAMPLER_H__
#define __GST_AUDIO_RESAMPLER_H__
#include <gst/gst.h>
#include <gst/audio/audio.h>
G_BEGIN_DECLS
typedef struct _GstAudioResampler GstAudioResampler;
/**
* GST_AUDIO_RESAMPLER_OPT_CUTOFF:
*
* G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_CUTOFF "GstAudioResampler.cutoff"
/**
* GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION:
*
* G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
* after the stopband for the kaiser window. 85 dB is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"
/**
* GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH:
*
* G_TYPE_DOUBLE, transition bandwidth. The width of the
* transition band for the kaiser window. 0.087 is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH "GstAudioResampler.transition-bandwidth"
/**
* GST_AUDIO_RESAMPLER_OPT_CUBIC_B:
*
* G_TYPE_DOUBLE, B parameter of the cubic filter.
* Values between 0.0 and 2.0 are accepted. 1.0 is the default.
*
* Below are some values of popular filters:
* B C
* Hermite 0.0 0.0
* Spline 1.0 0.0
* Catmull-Rom 0.0 1/2
*/
#define GST_AUDIO_RESAMPLER_OPT_CUBIC_B "GstAudioResampler.cubic-b"
/**
* GST_AUDIO_RESAMPLER_OPT_CUBIC_C:
*
* G_TYPE_DOUBLE, C parameter of the cubic filter.
* Values between 0.0 and 2.0 are accepted. 0.0 is the default.
*
* See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values
*/
#define GST_AUDIO_RESAMPLER_OPT_CUBIC_C "GstAudioResampler.cubic-c"
/**
* GST_AUDIO_RESAMPLER_OPT_N_TAPS:
*
* G_TYPE_INT: the number of taps to use for the filter.
* 0 is the default and selects the taps automatically.
*/
#define GST_AUDIO_RESAMPLER_OPT_N_TAPS "GstAudioResampler.n-taps"
/**
* GstAudioResamplerFilterMode:
* @GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED: Use interpolated filter tables. This
* uses less memory but more CPU and is slightly less accurate but it allows for more
* efficient variable rate resampling with gst_audio_resampler_update().
* @GST_AUDIO_RESAMPLER_FILTER_MODE_FULL: Use full filter table. This uses more memory
* but less CPU.
* @GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO: Automatically choose between interpolated
* and full filter tables.
*
* Select for the filter tables should be set up.
*/
typedef enum {
GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED = (0),
GST_AUDIO_RESAMPLER_FILTER_MODE_FULL,
GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO,
} GstAudioResamplerFilterMode;
/**
* GST_AUDIO_RESAMPLER_OPT_FILTER_MODE:
*
* GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
* constructed.
* GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE "GstAudioResampler.filter-mode"
/**
* GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD:
*
* G_TYPE_UINT: the amount of memory to use for full filter tables before
* switching to interpolated filter tables.
* 1048576 is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD "GstAudioResampler.filter-mode-threshold"
/**
* GstAudioResamplerFilterInterpolation:
* @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE: no interpolation
* @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: linear interpolation of the
* filter coeficients.
* @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: cubic interpolation of the
* filter coeficients.
*
* The different filter interpolation methods.
*/
typedef enum {
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE = (0),
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR,
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC,
} GstAudioResamplerFilterInterpolation;
/**
* GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION:
*
* GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be
* interpolated.
* GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.
*/
#define GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION "GstAudioResampler.filter-interpolation"
/**
* GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE:
*
* G_TYPE_UINT, oversampling to use when interpolating filters
* 8 is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE "GstAudioResampler.filter-oversample"
/**
* GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR:
*
* G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
* rates.
* 0.1 is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR "GstAudioResampler.max-phase-error"
/**
* GstAudioResamplerMethod:
* @GST_AUDIO_RESAMPLER_METHOD_NEAREST: Duplicates the samples when
* upsampling and drops when downsampling
* @GST_AUDIO_RESAMPLER_METHOD_LINEAR: Uses linear interpolation to reconstruct
* missing samples and averaging to downsample
* @GST_AUDIO_RESAMPLER_METHOD_CUBIC: Uses cubic interpolation
* @GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL: Uses Blackman-Nuttall windowed sinc interpolation
* @GST_AUDIO_RESAMPLER_METHOD_KAISER: Uses Kaiser windowed sinc interpolation
*
* Different subsampling and upsampling methods
*
* Since: 1.6
*/
typedef enum {
GST_AUDIO_RESAMPLER_METHOD_NEAREST,
GST_AUDIO_RESAMPLER_METHOD_LINEAR,
GST_AUDIO_RESAMPLER_METHOD_CUBIC,
GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL,
GST_AUDIO_RESAMPLER_METHOD_KAISER
} GstAudioResamplerMethod;
/**
* GstAudioResamplerFlags:
* @GST_AUDIO_RESAMPLER_FLAG_NONE: no flags
* @GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN: input samples are non-interleaved.
* an array of blocks of samples, one for each channel, should be passed to the
* resample function.
* @GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT: output samples are non-interleaved.
* an array of blocks of samples, one for each channel, should be passed to the
* resample function.
* @GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE: optimize for dynamic updates of the sample
* rates with gst_audio_resampler_update(). This will select an interpolating filter
* when #GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.
*
* Different resampler flags.
*/
typedef enum {
GST_AUDIO_RESAMPLER_FLAG_NONE = (0),
GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN = (1 << 0),
GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT = (1 << 1),
GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE = (1 << 2),
} GstAudioResamplerFlags;
#define GST_AUDIO_RESAMPLER_QUALITY_MIN 0
#define GST_AUDIO_RESAMPLER_QUALITY_MAX 10
#define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4
GST_AUDIO_API
void gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
guint quality,
gint in_rate, gint out_rate,
GstStructure *options);
GST_AUDIO_API
GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMethod method,
GstAudioResamplerFlags flags,
GstAudioFormat format, gint channels,
gint in_rate, gint out_rate,
GstStructure *options);
GST_AUDIO_API
void gst_audio_resampler_free (GstAudioResampler *resampler);
GST_AUDIO_API
void gst_audio_resampler_reset (GstAudioResampler *resampler);
GST_AUDIO_API
gboolean gst_audio_resampler_update (GstAudioResampler *resampler,
gint in_rate, gint out_rate,
GstStructure *options);
GST_AUDIO_API
gsize gst_audio_resampler_get_out_frames (GstAudioResampler *resampler,
gsize in_frames);
GST_AUDIO_API
gsize gst_audio_resampler_get_in_frames (GstAudioResampler *resampler,
gsize out_frames);
GST_AUDIO_API
gsize gst_audio_resampler_get_max_latency (GstAudioResampler *resampler);
GST_AUDIO_API
void gst_audio_resampler_resample (GstAudioResampler * resampler,
gpointer in[], gsize in_frames,
gpointer out[], gsize out_frames);
G_END_DECLS
#endif /* __GST_AUDIO_RESAMPLER_H__ */