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https://git.videolan.org/git/ffmpeg.git
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221 lines
6.5 KiB
C
221 lines
6.5 KiB
C
/*
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* Audio Processing Technology codec for Bluetooth (aptX)
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*
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* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
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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 AVCODEC_APTX_H
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#define AVCODEC_APTX_H
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "mathops.h"
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#include "audio_frame_queue.h"
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enum channels {
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LEFT,
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RIGHT,
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NB_CHANNELS
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};
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enum subbands {
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LF, // Low Frequency (0-5.5 kHz)
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MLF, // Medium-Low Frequency (5.5-11kHz)
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MHF, // Medium-High Frequency (11-16.5kHz)
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HF, // High Frequency (16.5-22kHz)
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NB_SUBBANDS
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};
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#define NB_FILTERS 2
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#define FILTER_TAPS 16
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typedef struct {
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int pos;
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int32_t buffer[2*FILTER_TAPS];
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} FilterSignal;
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typedef struct {
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FilterSignal outer_filter_signal[NB_FILTERS];
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FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS];
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} QMFAnalysis;
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typedef struct {
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int32_t quantized_sample;
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int32_t quantized_sample_parity_change;
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int32_t error;
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} Quantize;
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typedef struct {
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int32_t quantization_factor;
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int32_t factor_select;
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int32_t reconstructed_difference;
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} InvertQuantize;
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typedef struct {
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int32_t prev_sign[2];
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int32_t s_weight[2];
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int32_t d_weight[24];
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int32_t pos;
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int32_t reconstructed_differences[48];
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int32_t previous_reconstructed_sample;
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int32_t predicted_difference;
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int32_t predicted_sample;
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} Prediction;
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typedef struct {
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int32_t codeword_history;
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int32_t dither_parity;
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int32_t dither[NB_SUBBANDS];
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QMFAnalysis qmf;
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Quantize quantize[NB_SUBBANDS];
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InvertQuantize invert_quantize[NB_SUBBANDS];
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Prediction prediction[NB_SUBBANDS];
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} Channel;
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typedef struct {
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int hd;
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int block_size;
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int32_t sync_idx;
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Channel channels[NB_CHANNELS];
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AudioFrameQueue afq;
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} AptXContext;
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typedef const struct {
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const int32_t *quantize_intervals;
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const int32_t *invert_quantize_dither_factors;
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const int32_t *quantize_dither_factors;
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const int16_t *quantize_factor_select_offset;
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int tables_size;
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int32_t factor_max;
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int32_t prediction_order;
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} ConstTables;
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extern ConstTables ff_aptx_quant_tables[2][NB_SUBBANDS];
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/* Rounded right shift with optionnal clipping */
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#define RSHIFT_SIZE(size) \
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av_always_inline \
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static int##size##_t rshift##size(int##size##_t value, int shift) \
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{ \
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int##size##_t rounding = (int##size##_t)1 << (shift - 1); \
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int##size##_t mask = ((int##size##_t)1 << (shift + 1)) - 1; \
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return ((value + rounding) >> shift) - ((value & mask) == rounding); \
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} \
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av_always_inline \
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static int##size##_t rshift##size##_clip24(int##size##_t value, int shift) \
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{ \
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return av_clip_intp2(rshift##size(value, shift), 23); \
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}
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RSHIFT_SIZE(32)
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RSHIFT_SIZE(64)
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/*
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* Convolution filter coefficients for the outer QMF of the QMF tree.
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* The 2 sets are a mirror of each other.
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*/
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static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS] = {
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{
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730, -413, -9611, 43626, -121026, 269973, -585547, 2801966,
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697128, -160481, 27611, 8478, -10043, 3511, 688, -897,
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},
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{
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-897, 688, 3511, -10043, 8478, 27611, -160481, 697128,
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2801966, -585547, 269973, -121026, 43626, -9611, -413, 730,
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},
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};
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/*
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* Convolution filter coefficients for the inner QMF of the QMF tree.
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* The 2 sets are a mirror of each other.
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*/
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static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS] = {
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{
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1033, -584, -13592, 61697, -171156, 381799, -828088, 3962579,
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985888, -226954, 39048, 11990, -14203, 4966, 973, -1268,
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},
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{
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-1268, 973, 4966, -14203, 11990, 39048, -226954, 985888,
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3962579, -828088, 381799, -171156, 61697, -13592, -584, 1033,
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},
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};
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/*
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* Push one sample into a circular signal buffer.
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*/
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av_always_inline
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static void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample)
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{
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signal->buffer[signal->pos ] = sample;
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signal->buffer[signal->pos+FILTER_TAPS] = sample;
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signal->pos = (signal->pos + 1) & (FILTER_TAPS - 1);
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}
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/*
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* Compute the convolution of the signal with the coefficients, and reduce
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* to 24 bits by applying the specified right shifting.
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*/
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av_always_inline
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static int32_t aptx_qmf_convolution(FilterSignal *signal,
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const int32_t coeffs[FILTER_TAPS],
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int shift)
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{
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int32_t *sig = &signal->buffer[signal->pos];
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int64_t e = 0;
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int i;
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for (i = 0; i < FILTER_TAPS; i++)
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e += MUL64(sig[i], coeffs[i]);
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return rshift64_clip24(e, shift);
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}
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static inline int32_t aptx_quantized_parity(Channel *channel)
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{
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int32_t parity = channel->dither_parity;
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int subband;
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for (subband = 0; subband < NB_SUBBANDS; subband++)
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parity ^= channel->quantize[subband].quantized_sample;
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return parity & 1;
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}
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/* For each sample, ensure that the parity of all subbands of all channels
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* is 0 except once every 8 samples where the parity is forced to 1. */
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static inline int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx)
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{
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int32_t parity = aptx_quantized_parity(&channels[LEFT])
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^ aptx_quantized_parity(&channels[RIGHT]);
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int eighth = *idx == 7;
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*idx = (*idx + 1) & 7;
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return parity ^ eighth;
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}
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void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd);
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void ff_aptx_generate_dither(Channel *channel);
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int ff_aptx_init(AVCodecContext *avctx);
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#endif /* AVCODEC_APTX_H */
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