/* GStreamer * Copyright (C) <2015> Wim Taymans * * 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 #include 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__ */