/* 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__ */