/* GStreamer * Copyright (C) 2004 Ronald Bultje * (C) 2015 Wim Taymans * * audioconverter.h: audio format conversion library * * 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_CONVERTER_H__ #define __GST_AUDIO_CONVERTER_H__ #include #include G_BEGIN_DECLS typedef struct _GstAudioConverter GstAudioConverter; /** * GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD: * * #GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when * changing sample rates. * Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL. */ #define GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD "GstAudioConverter.resampler-method" /** * GST_AUDIO_CONVERTER_OPT_DITHER_METHOD: * * #GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when * changing bit depth. * Default is #GST_AUDIO_DITHER_NONE. */ #define GST_AUDIO_CONVERTER_OPT_DITHER_METHOD "GstAudioConverter.dither-method" /** * GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD: * * #GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use * to mask noise from quantization errors. * Default is #GST_AUDIO_NOISE_SHAPING_NONE. */ #define GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD "GstAudioConverter.noise-shaping-method" /** * GST_AUDIO_CONVERTER_OPT_QUANTIZATION: * * #G_TYPE_UINT, The quantization amount. Components will be * quantized to multiples of this value. * Default is 1 */ #define GST_AUDIO_CONVERTER_OPT_QUANTIZATION "GstAudioConverter.quantization" /** * GST_AUDIO_CONVERTER_OPT_MIX_MATRIX: * * #GST_TYPE_VALUE_LIST, The channel mapping matrix. * * The matrix coefficients must be between -1 and 1: the number of rows is equal * to the number of output channels and the number of columns is equal to the * number of input channels. * * ## Example matrix generation code * To generate the matrix using code: * * |[ * GValue v = G_VALUE_INIT; * GValue v2 = G_VALUE_INIT; * GValue v3 = G_VALUE_INIT; * * g_value_init (&v2, GST_TYPE_ARRAY); * g_value_init (&v3, G_TYPE_DOUBLE); * g_value_set_double (&v3, 1); * gst_value_array_append_value (&v2, &v3); * g_value_unset (&v3); * [ Repeat for as many double as your input channels - unset and reinit v3 ] * g_value_init (&v, GST_TYPE_ARRAY); * gst_value_array_append_value (&v, &v2); * g_value_unset (&v2); * [ Repeat for as many v2's as your output channels - unset and reinit v2] * g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v); * g_value_unset (&v); * ]| */ #define GST_AUDIO_CONVERTER_OPT_MIX_MATRIX "GstAudioConverter.mix-matrix" /** * GstAudioConverterFlags: * @GST_AUDIO_CONVERTER_FLAG_NONE: no flag * @GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE: the input sample arrays are writable and can be * used as temporary storage during conversion. * @GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE: allow arbitrary rate updates with * gst_audio_converter_update_config(). * * Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples(). */ typedef enum { GST_AUDIO_CONVERTER_FLAG_NONE = 0, GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE = (1 << 0), GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE = (1 << 1) } GstAudioConverterFlags; GST_AUDIO_API GstAudioConverter * gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo *in_info, GstAudioInfo *out_info, GstStructure *config); GST_AUDIO_API GType gst_audio_converter_get_type (void); GST_AUDIO_API void gst_audio_converter_free (GstAudioConverter * convert); GST_AUDIO_API void gst_audio_converter_reset (GstAudioConverter * convert); GST_AUDIO_API gboolean gst_audio_converter_update_config (GstAudioConverter * convert, gint in_rate, gint out_rate, GstStructure *config); GST_AUDIO_API const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert, gint *in_rate, gint *out_rate); GST_AUDIO_API gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert, gsize in_frames); GST_AUDIO_API gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert, gsize out_frames); GST_AUDIO_API gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert); GST_AUDIO_API gboolean gst_audio_converter_samples (GstAudioConverter * convert, GstAudioConverterFlags flags, gpointer in[], gsize in_frames, gpointer out[], gsize out_frames); GST_AUDIO_API gboolean gst_audio_converter_supports_inplace (GstAudioConverter *convert); GST_AUDIO_API gboolean gst_audio_converter_convert (GstAudioConverter * convert, GstAudioConverterFlags flags, gpointer in, gsize in_size, gpointer *out, gsize *out_size); G_END_DECLS #endif /* __GST_AUDIO_CONVERTER_H__ */