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099ac9faf2
Modernizing the documentation, making it simpler to read an modify and allowing us to possibly switch to hotdoc in the future.
235 lines
9.1 KiB
C
235 lines
9.1 KiB
C
/* GStreamer
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* Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_AUDIO_RESAMPLER_H__
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#define __GST_AUDIO_RESAMPLER_H__
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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typedef struct _GstAudioResampler GstAudioResampler;
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/**
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* GST_AUDIO_RESAMPLER_OPT_CUTOFF:
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*
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* G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_CUTOFF "GstAudioResampler.cutoff"
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/**
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* GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUTATION:
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*
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* G_TYPE_DOUBLE, stopband attenuation in debibels. The attenutation
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* after the stopband for the kaiser window. 85 dB is the default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"
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/**
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* GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH:
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*
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* G_TYPE_DOUBLE, transition bandwidth. The width of the
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* transition band for the kaiser window. 0.087 is the default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH "GstAudioResampler.transition-bandwidth"
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/**
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* GST_AUDIO_RESAMPLER_OPT_CUBIC_B:
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*
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* G_TYPE_DOUBLE, B parameter of the cubic filter.
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* Values between 0.0 and 2.0 are accepted. 1.0 is the default.
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*
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* Below are some values of popular filters:
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* B C
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* Hermite 0.0 0.0
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* Spline 1.0 0.0
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* Catmull-Rom 0.0 1/2
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*/
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#define GST_AUDIO_RESAMPLER_OPT_CUBIC_B "GstAudioResampler.cubic-b"
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/**
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* GST_AUDIO_RESAMPLER_OPT_CUBIC_C:
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*
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* G_TYPE_DOUBLE, C parameter of the cubic filter.
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* Values between 0.0 and 2.0 are accepted. 0.0 is the default.
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*
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* See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values
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*/
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#define GST_AUDIO_RESAMPLER_OPT_CUBIC_C "GstAudioResampler.cubic-c"
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/**
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* GST_AUDIO_RESAMPLER_OPT_N_TAPS:
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*
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* G_TYPE_INT: the number of taps to use for the filter.
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* 0 is the default and selects the taps automatically.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_N_TAPS "GstAudioResampler.n-taps"
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/**
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* GstAudioResamplerFilterMode:
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* @GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED: Use interpolated filter tables. This
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* uses less memory but more CPU and is slightly less accurate but it allows for more
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* efficient variable rate resampling with gst_audio_resampler_update().
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* @GST_AUDIO_RESAMPLER_FILTER_MODE_FULL: Use full filter table. This uses more memory
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* but less CPU.
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* @GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO: Automatically choose between interpolated
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* and full filter tables.
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*
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* Select for the filter tables should be set up.
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*/
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typedef enum {
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GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED = (0),
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GST_AUDIO_RESAMPLER_FILTER_MODE_FULL,
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GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO,
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} GstAudioResamplerFilterMode;
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/**
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* GST_AUDIO_RESAMPLER_OPT_FILTER_MODE:
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*
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* GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
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* constructed.
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* GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE "GstAudioResampler.filter-mode"
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/**
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* GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD:
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*
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* G_TYPE_UINT: the amount of memory to use for full filter tables before
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* switching to interpolated filter tables.
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* 1048576 is the default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD "GstAudioResampler.filter-mode-threshold"
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/**
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* GstAudioResamplerFilterInterpolation:
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* @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE: no interpolation
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* @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: linear interpolation of the
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* filter coeficients.
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* @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: cubic interpolation of the
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* filter coeficients.
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*
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* The different filter interpolation methods.
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*/
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typedef enum {
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GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE = (0),
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GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR,
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GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC,
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} GstAudioResamplerFilterInterpolation;
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/**
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* GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION:
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*
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* GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be
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* interpolated.
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* GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION "GstAudioResampler.filter-interpolation"
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/**
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* GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE:
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*
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* G_TYPE_UINT, oversampling to use when interpolating filters
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* 8 is the default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE "GstAudioResampler.filter-oversample"
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/**
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* GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR:
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*
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* G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
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* rates.
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* 0.1 is the default.
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*/
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#define GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR "GstAudioResampler.max-phase-error"
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/**
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* GstAudioResamplerMethod:
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* @GST_AUDIO_RESAMPLER_METHOD_NEAREST: Duplicates the samples when
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* upsampling and drops when downsampling
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* @GST_AUDIO_RESAMPLER_METHOD_LINEAR: Uses linear interpolation to reconstruct
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* missing samples and averaging to downsample
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* @GST_AUDIO_RESAMPLER_METHOD_CUBIC: Uses cubic interpolation
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* @GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL: Uses Blackman-Nuttall windowed sinc interpolation
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* @GST_AUDIO_RESAMPLER_METHOD_KAISER: Uses Kaiser windowed sinc interpolation
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*
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* Different subsampling and upsampling methods
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*
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* Since: 1.6
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*/
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typedef enum {
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GST_AUDIO_RESAMPLER_METHOD_NEAREST,
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GST_AUDIO_RESAMPLER_METHOD_LINEAR,
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GST_AUDIO_RESAMPLER_METHOD_CUBIC,
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GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL,
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GST_AUDIO_RESAMPLER_METHOD_KAISER
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} GstAudioResamplerMethod;
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/**
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* GstAudioResamplerFlags:
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* @GST_AUDIO_RESAMPLER_FLAG_NONE: no flags
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* @GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN: input samples are non-interleaved.
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* an array of blocks of samples, one for each channel, should be passed to the
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* resample function.
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* @GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT: output samples are non-interleaved.
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* an array of blocks of samples, one for each channel, should be passed to the
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* resample function.
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* @GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE: optimize for dynamic updates of the sample
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* rates with gst_audio_resampler_update(). This will select an interpolating filter
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* when #GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.
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*
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* Different resampler flags.
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*/
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typedef enum {
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GST_AUDIO_RESAMPLER_FLAG_NONE = (0),
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GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN = (1 << 0),
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GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT = (1 << 1),
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GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE = (1 << 2),
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} GstAudioResamplerFlags;
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#define GST_AUDIO_RESAMPLER_QUALITY_MIN 0
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#define GST_AUDIO_RESAMPLER_QUALITY_MAX 10
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#define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4
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void gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
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guint quality,
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gint in_rate, gint out_rate,
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GstStructure *options);
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GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMethod method,
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GstAudioResamplerFlags flags,
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GstAudioFormat format, gint channels,
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gint in_rate, gint out_rate,
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GstStructure *options);
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void gst_audio_resampler_free (GstAudioResampler *resampler);
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void gst_audio_resampler_reset (GstAudioResampler *resampler);
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gboolean gst_audio_resampler_update (GstAudioResampler *resampler,
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gint in_rate, gint out_rate,
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GstStructure *options);
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gsize gst_audio_resampler_get_out_frames (GstAudioResampler *resampler,
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gsize in_frames);
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gsize gst_audio_resampler_get_in_frames (GstAudioResampler *resampler,
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gsize out_frames);
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gsize gst_audio_resampler_get_max_latency (GstAudioResampler *resampler);
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void gst_audio_resampler_resample (GstAudioResampler * resampler,
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gpointer in[], gsize in_frames,
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gpointer out[], gsize out_frames);
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G_END_DECLS
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#endif /* __GST_AUDIO_RESAMPLER_H__ */
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