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1edb2c4242
audioconvert's passthrough status can no longer be determined strictly from input / output caps equality, as a mix-matrix can now be specified. We now call gst_base_transform_set_passthrough dynamically, based on the return from the new gst_audio_converter_is_passthrough() API, which takes the mix matrix into account.
173 lines
6.4 KiB
C
173 lines
6.4 KiB
C
/* GStreamer
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* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
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* (C) 2015 Wim Taymans <wim.taymans@gmail.com>
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*
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* audioconverter.h: audio format conversion library
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_AUDIO_CONVERTER_H__
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#define __GST_AUDIO_CONVERTER_H__
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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typedef struct _GstAudioConverter GstAudioConverter;
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/**
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* GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD:
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*
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* #GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
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* changing sample rates.
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* Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.
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*/
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#define GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD "GstAudioConverter.resampler-method"
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/**
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* GST_AUDIO_CONVERTER_OPT_DITHER_METHOD:
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*
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* #GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
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* changing bit depth.
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* Default is #GST_AUDIO_DITHER_NONE.
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*/
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#define GST_AUDIO_CONVERTER_OPT_DITHER_METHOD "GstAudioConverter.dither-method"
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/**
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* GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD:
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*
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* #GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
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* to mask noise from quantization errors.
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* Default is #GST_AUDIO_NOISE_SHAPING_NONE.
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*/
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#define GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD "GstAudioConverter.noise-shaping-method"
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/**
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* GST_AUDIO_CONVERTER_OPT_QUANTIZATION:
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*
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* #G_TYPE_UINT, The quantization amount. Components will be
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* quantized to multiples of this value.
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* Default is 1
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*/
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#define GST_AUDIO_CONVERTER_OPT_QUANTIZATION "GstAudioConverter.quantization"
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/**
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* GST_AUDIO_CONVERTER_OPT_MIX_MATRIX:
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*
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* #GST_TYPE_VALUE_LIST, The channel mapping matrix.
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*
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* The matrix coefficients must be between -1 and 1: the number of rows is equal
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* to the number of output channels and the number of columns is equal to the
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* number of input channels.
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*
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* ## Example matrix generation code
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* To generate the matrix using code:
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*
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* |[
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* GValue v = G_VALUE_INIT;
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* GValue v2 = G_VALUE_INIT;
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* GValue v3 = G_VALUE_INIT;
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*
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* g_value_init (&v2, GST_TYPE_ARRAY);
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* g_value_init (&v3, G_TYPE_DOUBLE);
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* g_value_set_double (&v3, 1);
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* gst_value_array_append_value (&v2, &v3);
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* g_value_unset (&v3);
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* [ Repeat for as many double as your input channels - unset and reinit v3 ]
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* g_value_init (&v, GST_TYPE_ARRAY);
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* gst_value_array_append_value (&v, &v2);
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* g_value_unset (&v2);
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* [ Repeat for as many v2's as your output channels - unset and reinit v2]
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* g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v);
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* g_value_unset (&v);
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* ]|
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*/
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#define GST_AUDIO_CONVERTER_OPT_MIX_MATRIX "GstAudioConverter.mix-matrix"
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/**
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* GstAudioConverterFlags:
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* @GST_AUDIO_CONVERTER_FLAG_NONE: no flag
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* @GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE: the input sample arrays are writable and can be
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* used as temporary storage during conversion.
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* @GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE: allow arbitrary rate updates with
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* gst_audio_converter_update_config().
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*
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* Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples().
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*/
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typedef enum {
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GST_AUDIO_CONVERTER_FLAG_NONE = 0,
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GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE = (1 << 0),
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GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE = (1 << 1)
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} GstAudioConverterFlags;
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GST_AUDIO_API
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GstAudioConverter * gst_audio_converter_new (GstAudioConverterFlags flags,
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GstAudioInfo *in_info,
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GstAudioInfo *out_info,
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GstStructure *config);
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GST_AUDIO_API
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GType gst_audio_converter_get_type (void);
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GST_AUDIO_API
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void gst_audio_converter_free (GstAudioConverter * convert);
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GST_AUDIO_API
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void gst_audio_converter_reset (GstAudioConverter * convert);
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GST_AUDIO_API
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gboolean gst_audio_converter_update_config (GstAudioConverter * convert,
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gint in_rate, gint out_rate,
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GstStructure *config);
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GST_AUDIO_API
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const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert,
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gint *in_rate, gint *out_rate);
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GST_AUDIO_API
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gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert,
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gsize in_frames);
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GST_AUDIO_API
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gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert,
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gsize out_frames);
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GST_AUDIO_API
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gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert);
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GST_AUDIO_API
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gboolean gst_audio_converter_samples (GstAudioConverter * convert,
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GstAudioConverterFlags flags,
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gpointer in[], gsize in_frames,
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gpointer out[], gsize out_frames);
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GST_AUDIO_API
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gboolean gst_audio_converter_supports_inplace (GstAudioConverter *convert);
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GST_AUDIO_API
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gboolean gst_audio_converter_is_passthrough (GstAudioConverter *convert);
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GST_AUDIO_API
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gboolean gst_audio_converter_convert (GstAudioConverter * convert,
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GstAudioConverterFlags flags,
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gpointer in, gsize in_size,
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gpointer *out, gsize *out_size);
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G_END_DECLS
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#endif /* __GST_AUDIO_CONVERTER_H__ */
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