| /* GStreamer |
| * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net> |
| * (C) 2015 Wim Taymans <wim.taymans@gmail.com> |
| * |
| * 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 <gst/gst.h> |
| #include <gst/audio/audio.h> |
| |
| 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__ */ |