| /* 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__ */ |
