mirror of
https://github.com/FFmpeg/FFmpeg.git
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Kacper Michajłow
72b81d5b64
Convert DSD to PCM with the same 96-tap symmetric lowpass filter the libavcodec DSD decoders use. The implementation is copied from libavcodec/dsd.c, minus the LSBF table variants, since AV_SAMPLE_FMT_DSD is defined as MSBF. The libavcodec copy will be removed in future commits as decoders are transitioned to SWR. The conversion is stateful (per-channel filter history), so the audio conversion functions gain a per-channel state argument. DSD silence is the 0x69 bit pattern instead of the 0x80 used for PCM. Only a direct DSD to float conversion function is provided. For format pairs without a direct conversion function (e.g. DSD to s16 with equal rates), swr_init() now falls back from the full_convert shortcut to the generic two-stage path instead of failing, and reports unsupported conversions with a proper error message instead of ENOMEM. Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
80 lines
2.7 KiB
C
80 lines
2.7 KiB
C
/*
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* audio conversion
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* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2008 Peter Ross
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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#ifndef SWRESAMPLE_AUDIOCONVERT_H
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#define SWRESAMPLE_AUDIOCONVERT_H
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/**
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* @file
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* Audio format conversion routines
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*/
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#include "swresample_internal.h"
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#include "dsd2pcm.h"
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typedef void (conv_func_type)(DSDContext *st, uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end);
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typedef void (simd_func_type)(uint8_t **dst, const uint8_t **src, int len);
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typedef struct AudioConvert {
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int channels;
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int in_simd_align_mask;
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int out_simd_align_mask;
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conv_func_type *conv_f;
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simd_func_type *simd_f;
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const int *ch_map;
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uint8_t silence[8]; ///< silence input sample
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DSDContext dsd_state[SWR_CH_MAX]; ///< per-channel state for DSD input
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}AudioConvert;
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/**
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* Create an audio sample format converter context
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* @param out_fmt Output sample format
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* @param in_fmt Input sample format
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* @param channels Number of channels
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* @param flags See AV_CPU_FLAG_xx
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* @param ch_map list of the channels id to pick from the source stream, NULL
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* if all channels must be selected
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* @return NULL on error
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*/
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AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt,
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enum AVSampleFormat in_fmt,
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int channels, const int *ch_map,
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int flags);
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/**
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* Free audio sample format converter context.
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* and set the pointer to NULL
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*/
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void swri_audio_convert_free(AudioConvert **ctx);
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/**
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* Convert between audio sample formats
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* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
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* @param[in] in array of input buffers for each channel
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* @param len length of audio frame size (measured in samples)
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*/
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int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len);
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#endif /* SWRESAMPLE_AUDIOCONVERT_H */
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