This is a new library for audio sample format, channel layout, and sample rate conversion.
		
			
				
	
	
		
			90 lines
		
	
	
		
			5.3 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			90 lines
		
	
	
		
			5.3 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav 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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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "avresample.h"
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#include "internal.h"
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#include "audio_mix.h"
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/**
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 * @file
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 * Options definition for AVAudioResampleContext.
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 */
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#define OFFSET(x) offsetof(AVAudioResampleContext, x)
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#define PARAM AV_OPT_FLAG_AUDIO_PARAM
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static const AVOption options[] = {
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    { "in_channel_layout",      "Input Channel Layout",     OFFSET(in_channel_layout),      AV_OPT_TYPE_INT64,  { 0                     }, INT64_MIN,            INT64_MAX,              PARAM },
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    { "in_sample_fmt",          "Input Sample Format",      OFFSET(in_sample_fmt),          AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_S16     }, AV_SAMPLE_FMT_U8,     AV_SAMPLE_FMT_NB-1,     PARAM },
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    { "in_sample_rate",         "Input Sample Rate",        OFFSET(in_sample_rate),         AV_OPT_TYPE_INT,    { 48000                 }, 1,                    INT_MAX,                PARAM },
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    { "out_channel_layout",     "Output Channel Layout",    OFFSET(out_channel_layout),     AV_OPT_TYPE_INT64,  { 0                     }, INT64_MIN,            INT64_MAX,              PARAM },
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    { "out_sample_fmt",         "Output Sample Format",     OFFSET(out_sample_fmt),         AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_S16     }, AV_SAMPLE_FMT_U8,     AV_SAMPLE_FMT_NB-1,     PARAM },
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    { "out_sample_rate",        "Output Sample Rate",       OFFSET(out_sample_rate),        AV_OPT_TYPE_INT,    { 48000                 }, 1,                    INT_MAX,                PARAM },
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    { "internal_sample_fmt",    "Internal Sample Format",   OFFSET(internal_sample_fmt),    AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_FLTP    }, AV_SAMPLE_FMT_NONE,   AV_SAMPLE_FMT_NB-1,     PARAM },
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    { "mix_coeff_type",         "Mixing Coefficient Type",  OFFSET(mix_coeff_type),         AV_OPT_TYPE_INT,    { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q6, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
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        { "q6",  "16-bit 10.6 Fixed-Point",  0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q6  }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
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        { "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
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        { "flt", "Floating-Point",           0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
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    { "center_mix_level",       "Center Mix Level",         OFFSET(center_mix_level),       AV_OPT_TYPE_DOUBLE, { M_SQRT1_2             }, -32.0,                32.0,                   PARAM },
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    { "surround_mix_level",     "Surround Mix Level",       OFFSET(surround_mix_level),     AV_OPT_TYPE_DOUBLE, { M_SQRT1_2             }, -32.0,                32.0,                   PARAM },
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    { "lfe_mix_level",          "LFE Mix Level",            OFFSET(lfe_mix_level),          AV_OPT_TYPE_DOUBLE, { 0.0                   }, -32.0,                32.0,                   PARAM },
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    { "force_resampling",       "Force Resampling",         OFFSET(force_resampling),       AV_OPT_TYPE_INT,    { 0                     }, 0,                    1,                      PARAM },
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    { "filter_size",            "Resampling Filter Size",   OFFSET(filter_size),            AV_OPT_TYPE_INT,    { 16                    }, 0,                    32, /* ??? */           PARAM },
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    { "phase_shift",            "Resampling Phase Shift",   OFFSET(phase_shift),            AV_OPT_TYPE_INT,    { 10                    }, 0,                    30, /* ??? */           PARAM },
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    { "linear_interp",          "Use Linear Interpolation", OFFSET(linear_interp),          AV_OPT_TYPE_INT,    { 0                     }, 0,                    1,                      PARAM },
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    { "cutoff",                 "Cutoff Frequency Ratio",   OFFSET(cutoff),                 AV_OPT_TYPE_DOUBLE, { 0.8                   }, 0.0,                  1.0,                    PARAM },
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    { NULL },
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};
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static const AVClass av_resample_context_class = {
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    .class_name = "AVAudioResampleContext",
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    .item_name  = av_default_item_name,
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    .option     = options,
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    .version    = LIBAVUTIL_VERSION_INT,
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};
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AVAudioResampleContext *avresample_alloc_context(void)
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{
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    AVAudioResampleContext *avr;
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    avr = av_mallocz(sizeof(*avr));
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    if (!avr)
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        return NULL;
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    avr->av_class = &av_resample_context_class;
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    av_opt_set_defaults(avr);
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    avr->am = av_mallocz(sizeof(*avr->am));
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    if (!avr->am) {
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        av_free(avr);
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        return NULL;
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    }
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    avr->am->avr = avr;
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    return avr;
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}
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const AVClass *avresample_get_class(void)
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{
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    return &av_resample_context_class;
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}
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