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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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de37491662
Remove the consumed/produced output fields from the resampler and converter. Let the caler specify the right number of input/output samples so we can be more optimal. Use just one function to update the converter configuration. Simplify some things internally. Make it possible to use writable input as temp space in audioconvert.
185 lines
7.1 KiB
C
185 lines
7.1 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.
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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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* 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: samples are non-interleaved. an array
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* of blocks of samples, one for each channel, should be passed to the resample
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* function.
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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 = (1 << 0),
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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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guint in_rate, guint 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, guint channels,
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guint in_rate, guint out_rate,
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GstStructure *options);
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void gst_audio_resampler_free (GstAudioResampler *resampler);
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gboolean gst_audio_resampler_update (GstAudioResampler *resampler,
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guint in_rate, guint 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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