/* 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_ATTENUTATION * * G_TYPE_DOUBLE, stopband attenuation in debibels. The attenutation * 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. * @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" /** * 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: samples are non-interleaved. an array * of blocks of samples, one for each channel, should be passed to the resample * function. * * Different resampler flags. */ typedef enum { GST_AUDIO_RESAMPLER_FLAG_NONE = (0), GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED = (1 << 0), } GstAudioResamplerFlags; #define GST_AUDIO_RESAMPLER_QUALITY_MIN 0 #define GST_AUDIO_RESAMPLER_QUALITY_MAX 10 #define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4 void gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method, guint quality, guint in_rate, guint out_rate, GstStructure *options); GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMethod method, GstAudioResamplerFlags flags, GstAudioFormat format, guint channels, guint in_rate, guint out_rate, GstStructure *options); void gst_audio_resampler_free (GstAudioResampler *resampler); gboolean gst_audio_resampler_update (GstAudioResampler *resampler, guint in_rate, guint out_rate, GstStructure *options); gsize gst_audio_resampler_get_out_frames (GstAudioResampler *resampler, gsize in_frames); gsize gst_audio_resampler_get_in_frames (GstAudioResampler *resampler, gsize out_frames); gsize gst_audio_resampler_get_max_latency (GstAudioResampler *resampler); void gst_audio_resampler_resample (GstAudioResampler * resampler, gpointer in[], gsize in_frames, gpointer out[], gsize out_frames, gsize *in_consumed, gsize *out_produced); G_END_DECLS #endif /* __GST_AUDIO_RESAMPLER_H__ */