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010b9547d3
Optimize LE<->BE conversion by adding a dedicated fast path instead of using the generic converter. Implement transform_ip function in order to do the endian swap in place. This saves buffer allocation for the intermediate format, can be done in place and also performs the conversion in one step instead of unpack-convert-pack. For all bit widths the naive algorithm is implemented, which provides the best performance when compiled with -O3. ORC was considered but eventually removed as it requires a dedicated function for in-place conversion (due to the "restrict" parameters). A more complex algorithm for the 24-bit conversion with unrolled loop and 32-bit processing is implemented in the #if 0 section. It performs better if compiled with -O2. With -O3 however the naive algorithm performs better. https://bugzilla.gnome.org/show_bug.cgi?id=773073
113 lines
4.6 KiB
C
113 lines
4.6 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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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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* 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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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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void gst_audio_converter_free (GstAudioConverter * convert);
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void gst_audio_converter_reset (GstAudioConverter * convert);
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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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const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert,
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gint *in_rate, gint *out_rate);
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gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert,
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gsize in_frames);
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gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert,
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gsize out_frames);
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gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert);
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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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gboolean gst_audio_converter_supports_inplace (GstAudioConverter *convert);
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#endif /* __GST_AUDIO_CONVERTER_H__ */
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