avcodec/aptx: split decoder and encoder into separate files

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

libavcodec/Makefile

@@ -194,10 +194,10 @@ OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpegenc_common.o \
 OBJS-$(CONFIG_ANM_DECODER)             += anm.o
 OBJS-$(CONFIG_ANSI_DECODER)            += ansi.o cga_data.o
 OBJS-$(CONFIG_APE_DECODER)             += apedec.o
-OBJS-$(CONFIG_APTX_DECODER)            += aptx.o
-OBJS-$(CONFIG_APTX_ENCODER)            += aptx.o
-OBJS-$(CONFIG_APTX_HD_DECODER)         += aptx.o
-OBJS-$(CONFIG_APTX_HD_ENCODER)         += aptx.o
+OBJS-$(CONFIG_APTX_DECODER)            += aptxdec.o aptx.o
+OBJS-$(CONFIG_APTX_ENCODER)            += aptxenc.o aptx.o
+OBJS-$(CONFIG_APTX_HD_DECODER)         += aptxdec.o aptx.o
+OBJS-$(CONFIG_APTX_HD_ENCODER)         += aptxenc.o aptx.o
 OBJS-$(CONFIG_APNG_DECODER)            += png.o pngdec.o pngdsp.o
 OBJS-$(CONFIG_APNG_ENCODER)            += png.o pngenc.o
 OBJS-$(CONFIG_ARBC_DECODER)            += arbc.o

libavcodec/aptx.h

+/*
+ * 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 */

libavcodec/aptxdec.c

+/*
+ * 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
+ */
+
+#include "aptx.h"
+
+/*
+ * Half-band QMF synthesis filter realized with a polyphase FIR filter.
+ * Join 2 subbands and upsample by 2.
+ * So for each 2 subbands sample that goes in, a pair of samples goes out.
+ */
+av_always_inline
+static void aptx_qmf_polyphase_synthesis(FilterSignal signal[NB_FILTERS],
+                                         const int32_t coeffs[NB_FILTERS][FILTER_TAPS],
+                                         int shift,
+                                         int32_t low_subband_input,
+                                         int32_t high_subband_input,
+                                         int32_t samples[NB_FILTERS])
+{
+    int32_t subbands[NB_FILTERS];
+    int i;
+
+    subbands[0] = low_subband_input + high_subband_input;
+    subbands[1] = low_subband_input - high_subband_input;
+
+    for (i = 0; i < NB_FILTERS; i++) {
+        aptx_qmf_filter_signal_push(&signal[i], subbands[1-i]);
+        samples[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift);
+    }
+}
+
+/*
+ * Two stage QMF synthesis tree.
+ * Join 4 subbands and upsample by 4.
+ * So for each 4 subbands sample that goes in, a group of 4 samples goes out.
+ */
+static void aptx_qmf_tree_synthesis(QMFAnalysis *qmf,
+                                    int32_t subband_samples[4],
+                                    int32_t samples[4])
+{
+    int32_t intermediate_samples[4];
+    int i;
+
+    /* Join 4 subbands into 2 intermediate subbands upsampled to 2 samples. */
+    for (i = 0; i < 2; i++)
+        aptx_qmf_polyphase_synthesis(qmf->inner_filter_signal[i],
+                                     aptx_qmf_inner_coeffs, 22,
+                                     subband_samples[2*i+0],
+                                     subband_samples[2*i+1],
+                                     &intermediate_samples[2*i]);
+
+    /* Join 2 samples from intermediate subbands upsampled to 4 samples. */
+    for (i = 0; i < 2; i++)
+        aptx_qmf_polyphase_synthesis(qmf->outer_filter_signal,
+                                     aptx_qmf_outer_coeffs, 21,
+                                     intermediate_samples[0+i],
+                                     intermediate_samples[2+i],
+                                     &samples[2*i]);
+}
+
+
+static void aptx_decode_channel(Channel *channel, int32_t samples[4])
+{
+    int32_t subband_samples[4];
+    int subband;
+    for (subband = 0; subband < NB_SUBBANDS; subband++)
+        subband_samples[subband] = channel->prediction[subband].previous_reconstructed_sample;
+    aptx_qmf_tree_synthesis(&channel->qmf, subband_samples, samples);
+}
+
+static void aptx_unpack_codeword(Channel *channel, uint16_t codeword)
+{
+    channel->quantize[0].quantized_sample = sign_extend(codeword >>  0, 7);
+    channel->quantize[1].quantized_sample = sign_extend(codeword >>  7, 4);
+    channel->quantize[2].quantized_sample = sign_extend(codeword >> 11, 2);
+    channel->quantize[3].quantized_sample = sign_extend(codeword >> 13, 3);
+    channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
+                                          | aptx_quantized_parity(channel);
+}
+
+static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword)
+{
+    channel->quantize[0].quantized_sample = sign_extend(codeword >>  0, 9);
+    channel->quantize[1].quantized_sample = sign_extend(codeword >>  9, 6);
+    channel->quantize[2].quantized_sample = sign_extend(codeword >> 15, 4);
+    channel->quantize[3].quantized_sample = sign_extend(codeword >> 19, 5);
+    channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1)
+                                          | aptx_quantized_parity(channel);
+}
+
+static int aptx_decode_samples(AptXContext *ctx,
+                                const uint8_t *input,
+                                int32_t samples[NB_CHANNELS][4])
+{
+    int channel, ret;
+
+    for (channel = 0; channel < NB_CHANNELS; channel++) {
+        ff_aptx_generate_dither(&ctx->channels[channel]);
+
+        if (ctx->hd)
+            aptxhd_unpack_codeword(&ctx->channels[channel],
+                                   AV_RB24(input + 3*channel));
+        else
+            aptx_unpack_codeword(&ctx->channels[channel],
+                                 AV_RB16(input + 2*channel));
+        ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd);
+    }
+
+    ret = aptx_check_parity(ctx->channels, &ctx->sync_idx);
+
+    for (channel = 0; channel < NB_CHANNELS; channel++)
+        aptx_decode_channel(&ctx->channels[channel], samples[channel]);
+
+    return ret;
+}
+
+static int aptx_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    AptXContext *s = avctx->priv_data;
+    AVFrame *frame = data;
+    int pos, opos, channel, sample, ret;
+
+    if (avpkt->size < s->block_size) {
+        av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* get output buffer */
+    frame->channels = NB_CHANNELS;
+    frame->format = AV_SAMPLE_FMT_S32P;
+    frame->nb_samples = 4 * avpkt->size / s->block_size;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+
+    for (pos = 0, opos = 0; opos < frame->nb_samples; pos += s->block_size, opos += 4) {
+        int32_t samples[NB_CHANNELS][4];
+
+        if (aptx_decode_samples(s, &avpkt->data[pos], samples)) {
+            av_log(avctx, AV_LOG_ERROR, "Synchronization error\n");
+            return AVERROR_INVALIDDATA;
+        }
+
+        for (channel = 0; channel < NB_CHANNELS; channel++)
+            for (sample = 0; sample < 4; sample++)
+                AV_WN32A(&frame->data[channel][4*(opos+sample)],
+                         samples[channel][sample] * 256);
+    }
+
+    *got_frame_ptr = 1;
+    return s->block_size * frame->nb_samples / 4;
+}
+
+#if CONFIG_APTX_DECODER
+AVCodec ff_aptx_decoder = {
+    .name                  = "aptx",
+    .long_name             = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"),
+    .type                  = AVMEDIA_TYPE_AUDIO,
+    .id                    = AV_CODEC_ID_APTX,
+    .priv_data_size        = sizeof(AptXContext),
+    .init                  = ff_aptx_init,
+    .decode                = aptx_decode_frame,
+    .capabilities          = AV_CODEC_CAP_DR1,
+    .caps_internal         = FF_CODEC_CAP_INIT_THREADSAFE,
+    .channel_layouts       = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
+    .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
+                                                             AV_SAMPLE_FMT_NONE },
+};
+#endif
+
+#if CONFIG_APTX_HD_DECODER
+AVCodec ff_aptx_hd_decoder = {
+    .name                  = "aptx_hd",
+    .long_name             = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"),
+    .type                  = AVMEDIA_TYPE_AUDIO,
+    .id                    = AV_CODEC_ID_APTX_HD,
+    .priv_data_size        = sizeof(AptXContext),
+    .init                  = ff_aptx_init,
+    .decode                = aptx_decode_frame,
+    .capabilities          = AV_CODEC_CAP_DR1,
+    .caps_internal         = FF_CODEC_CAP_INIT_THREADSAFE,
+    .channel_layouts       = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0},
+    .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
+                                                             AV_SAMPLE_FMT_NONE },
+};
+#endif