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/*
* Audio Processing Technology codec for Bluetooth (aptX)
*
* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_APTX_H
#define AVCODEC_APTX_H
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "mathops.h"
#include "audio_frame_queue.h"
enum channels {
LEFT,
RIGHT,
NB_CHANNELS
};
enum subbands {
LF, // Low Frequency (0-5.5 kHz)
MLF, // Medium-Low Frequency (5.5-11kHz)
MHF, // Medium-High Frequency (11-16.5kHz)
HF, // High Frequency (16.5-22kHz)
NB_SUBBANDS
};
#define NB_FILTERS 2
#define FILTER_TAPS 16
typedef struct {
int pos;
int32_t buffer[2*FILTER_TAPS];
} FilterSignal;
typedef struct {
FilterSignal outer_filter_signal[NB_FILTERS];
FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS];
} QMFAnalysis;
typedef struct {
int32_t quantized_sample;
int32_t quantized_sample_parity_change;
int32_t error;
} Quantize;
typedef struct {
int32_t quantization_factor;
int32_t factor_select;
int32_t reconstructed_difference;
} InvertQuantize;
typedef struct {
int32_t prev_sign[2];
int32_t s_weight[2];
int32_t d_weight[24];
int32_t pos;
int32_t reconstructed_differences[48];
int32_t previous_reconstructed_sample;
int32_t predicted_difference;
int32_t predicted_sample;
} Prediction;
typedef struct {
int32_t codeword_history;
int32_t dither_parity;
int32_t dither[NB_SUBBANDS];
QMFAnalysis qmf;
Quantize quantize[NB_SUBBANDS];
InvertQuantize invert_quantize[NB_SUBBANDS];
Prediction prediction[NB_SUBBANDS];
} Channel;
typedef struct {
int hd;
int block_size;
int32_t sync_idx;
Channel channels[NB_CHANNELS];
AudioFrameQueue afq;
} AptXContext;
typedef const struct {
const int32_t *quantize_intervals;
const int32_t *invert_quantize_dither_factors;
const int32_t *quantize_dither_factors;
const int16_t *quantize_factor_select_offset;
int tables_size;
int32_t factor_max;
int32_t prediction_order;
} ConstTables;
extern ConstTables ff_aptx_quant_tables[2][NB_SUBBANDS];
/* Rounded right shift with optionnal clipping */
#define RSHIFT_SIZE(size) \
av_always_inline \
static int##size##_t rshift##size(int##size##_t value, int shift) \
{ \
int##size##_t rounding = (int##size##_t)1 << (shift - 1); \
int##size##_t mask = ((int##size##_t)1 << (shift + 1)) - 1; \
return ((value + rounding) >> shift) - ((value & mask) == rounding); \
} \
av_always_inline \
static int##size##_t rshift##size##_clip24(int##size##_t value, int shift) \
{ \
return av_clip_intp2(rshift##size(value, shift), 23); \
}
RSHIFT_SIZE(32)
RSHIFT_SIZE(64)
/*
* Convolution filter coefficients for the outer QMF of the QMF tree.
* The 2 sets are a mirror of each other.
*/
static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS] = {
{
730, -413, -9611, 43626, -121026, 269973, -585547, 2801966,
697128, -160481, 27611, 8478, -10043, 3511, 688, -897,
},
{
-897, 688, 3511, -10043, 8478, 27611, -160481, 697128,
2801966, -585547, 269973, -121026, 43626, -9611, -413, 730,
},
};
/*
* Convolution filter coefficients for the inner QMF of the QMF tree.
* The 2 sets are a mirror of each other.
*/
static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS] = {
{
1033, -584, -13592, 61697, -171156, 381799, -828088, 3962579,
985888, -226954, 39048, 11990, -14203, 4966, 973, -1268,
},
{
-1268, 973, 4966, -14203, 11990, 39048, -226954, 985888,
3962579, -828088, 381799, -171156, 61697, -13592, -584, 1033,
},
};
/*
* Push one sample into a circular signal buffer.
*/
av_always_inline
static void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample)
{
signal->buffer[signal->pos ] = sample;
signal->buffer[signal->pos+FILTER_TAPS] = sample;
signal->pos = (signal->pos + 1) & (FILTER_TAPS - 1);
}
/*
* Compute the convolution of the signal with the coefficients, and reduce
* to 24 bits by applying the specified right shifting.
*/
av_always_inline
static int32_t aptx_qmf_convolution(FilterSignal *signal,
const int32_t coeffs[FILTER_TAPS],
int shift)
{
int32_t *sig = &signal->buffer[signal->pos];
int64_t e = 0;
int i;
for (i = 0; i < FILTER_TAPS; i++)
e += MUL64(sig[i], coeffs[i]);
return rshift64_clip24(e, shift);
}
static inline int32_t aptx_quantized_parity(Channel *channel)
{
int32_t parity = channel->dither_parity;
int subband;
for (subband = 0; subband < NB_SUBBANDS; subband++)
parity ^= channel->quantize[subband].quantized_sample;
return parity & 1;
}
/* For each sample, ensure that the parity of all subbands of all channels
* is 0 except once every 8 samples where the parity is forced to 1. */
static inline int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx)
{
int32_t parity = aptx_quantized_parity(&channels[LEFT])
^ aptx_quantized_parity(&channels[RIGHT]);
int eighth = *idx == 7;
*idx = (*idx + 1) & 7;
return parity ^ eighth;
}
void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd);
void ff_aptx_generate_dither(Channel *channel);
int ff_aptx_init(AVCodecContext *avctx);
#endif /* AVCODEC_APTX_H */
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