avcodec/aptx: split decoder and encoder into separate files

Signed-off-by: James Almer <jamrial@gmail.com>

1 files changed, 220 insertions, 0 deletions

diff --git a/libavcodec/aptx.h b/libavcodec/aptx.h
new file mode 100644
index 0000000000..ce3d7dc6c1
--- /dev/null
+++ b/libavcodec/aptx.h
@@ -0,0 +1,220 @@
+/*
+ * 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 "internal.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 */