mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 02:30:35 +00:00
246 lines
9.2 KiB
C
246 lines
9.2 KiB
C
/* 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_EXPORT
|
|
void gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
|
|
guint quality,
|
|
gint in_rate, gint out_rate,
|
|
GstStructure *options);
|
|
|
|
GST_EXPORT
|
|
GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMethod method,
|
|
GstAudioResamplerFlags flags,
|
|
GstAudioFormat format, gint channels,
|
|
gint in_rate, gint out_rate,
|
|
GstStructure *options);
|
|
|
|
GST_EXPORT
|
|
void gst_audio_resampler_free (GstAudioResampler *resampler);
|
|
|
|
GST_EXPORT
|
|
void gst_audio_resampler_reset (GstAudioResampler *resampler);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_audio_resampler_update (GstAudioResampler *resampler,
|
|
gint in_rate, gint out_rate,
|
|
GstStructure *options);
|
|
|
|
GST_EXPORT
|
|
gsize gst_audio_resampler_get_out_frames (GstAudioResampler *resampler,
|
|
gsize in_frames);
|
|
|
|
GST_EXPORT
|
|
gsize gst_audio_resampler_get_in_frames (GstAudioResampler *resampler,
|
|
gsize out_frames);
|
|
|
|
GST_EXPORT
|
|
gsize gst_audio_resampler_get_max_latency (GstAudioResampler *resampler);
|
|
|
|
GST_EXPORT
|
|
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__ */
|