aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/aptxenc.c
blob: 5ea6053c26f38aed596abb35d47b1075f384df39 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
/*
 * 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 "libavutil/channel_layout.h"
#include "aptx.h"
#include "encode.h"

/*
 * Half-band QMF analysis filter realized with a polyphase FIR filter.
 * Split into 2 subbands and downsample by 2.
 * So for each pair of samples that goes in, one sample goes out,
 * split into 2 separate subbands.
 */
av_always_inline
static void aptx_qmf_polyphase_analysis(FilterSignal signal[NB_FILTERS],
                                        const int32_t coeffs[NB_FILTERS][FILTER_TAPS],
                                        int shift,
                                        int32_t samples[NB_FILTERS],
                                        int32_t *low_subband_output,
                                        int32_t *high_subband_output)
{
    int32_t subbands[NB_FILTERS];
    int i;

    for (i = 0; i < NB_FILTERS; i++) {
        aptx_qmf_filter_signal_push(&signal[i], samples[NB_FILTERS-1-i]);
        subbands[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift);
    }

    *low_subband_output  = av_clip_intp2(subbands[0] + subbands[1], 23);
    *high_subband_output = av_clip_intp2(subbands[0] - subbands[1], 23);
}

/*
 * Two stage QMF analysis tree.
 * Split 4 input samples into 4 subbands and downsample by 4.
 * So for each group of 4 samples that goes in, one sample goes out,
 * split into 4 separate subbands.
 */
static void aptx_qmf_tree_analysis(QMFAnalysis *qmf,
                                   int32_t samples[4],
                                   int32_t subband_samples[4])
{
    int32_t intermediate_samples[4];
    int i;

    /* Split 4 input samples into 2 intermediate subbands downsampled to 2 samples */
    for (i = 0; i < 2; i++)
        aptx_qmf_polyphase_analysis(qmf->outer_filter_signal,
                                    aptx_qmf_outer_coeffs, 23,
                                    &samples[2*i],
                                    &intermediate_samples[0+i],
                                    &intermediate_samples[2+i]);

    /* Split 2 intermediate subband samples into 4 final subbands downsampled to 1 sample */
    for (i = 0; i < 2; i++)
        aptx_qmf_polyphase_analysis(qmf->inner_filter_signal[i],
                                    aptx_qmf_inner_coeffs, 23,
                                    &intermediate_samples[2*i],
                                    &subband_samples[2*i+0],
                                    &subband_samples[2*i+1]);
}

av_always_inline
static int32_t aptx_bin_search(int32_t value, int32_t factor,
                               const int32_t *intervals, int32_t nb_intervals)
{
    int32_t idx = 0;
    int i;

    for (i = nb_intervals >> 1; i > 0; i >>= 1)
        if (MUL64(factor, intervals[idx + i]) <= ((int64_t)value << 24))
            idx += i;

    return idx;
}

static void aptx_quantize_difference(Quantize *quantize,
                                     int32_t sample_difference,
                                     int32_t dither,
                                     int32_t quantization_factor,
                                     ConstTables *tables)
{
    const int32_t *intervals = tables->quantize_intervals;
    int32_t quantized_sample, dithered_sample, parity_change;
    int32_t d, mean, interval, inv, sample_difference_abs;
    int64_t error;

    sample_difference_abs = FFABS(sample_difference);
    sample_difference_abs = FFMIN(sample_difference_abs, (1 << 23) - 1);

    quantized_sample = aptx_bin_search(sample_difference_abs >> 4,
                                       quantization_factor,
                                       intervals, tables->tables_size);

    d = rshift32_clip24(MULH(dither, dither), 7) - (1 << 23);
    d = rshift64(MUL64(d, tables->quantize_dither_factors[quantized_sample]), 23);

    intervals += quantized_sample;
    mean = (intervals[1] + intervals[0]) / 2;
    interval = (intervals[1] - intervals[0]) * (-(sample_difference < 0) | 1);

    dithered_sample = rshift64_clip24(MUL64(dither, interval) + ((int64_t)av_clip_intp2(mean + d, 23) << 32), 32);
    error = ((int64_t)sample_difference_abs << 20) - MUL64(dithered_sample, quantization_factor);
    quantize->error = FFABS(rshift64(error, 23));

    parity_change = quantized_sample;
    if (error < 0)
        quantized_sample--;
    else
        parity_change--;

    inv = -(sample_difference < 0);
    quantize->quantized_sample               = quantized_sample ^ inv;
    quantize->quantized_sample_parity_change = parity_change    ^ inv;
}

static void aptx_encode_channel(Channel *channel, int32_t samples[4], int hd)
{
    int32_t subband_samples[4];
    int subband;
    aptx_qmf_tree_analysis(&channel->qmf, samples, subband_samples);
    ff_aptx_generate_dither(channel);
    for (subband = 0; subband < NB_SUBBANDS; subband++) {
        int32_t diff = av_clip_intp2(subband_samples[subband] - channel->prediction[subband].predicted_sample, 23);
        aptx_quantize_difference(&channel->quantize[subband], diff,
                                 channel->dither[subband],
                                 channel->invert_quantize[subband].quantization_factor,
                                 &ff_aptx_quant_tables[hd][subband]);
    }
}

static void aptx_insert_sync(Channel channels[NB_CHANNELS], int32_t *idx)
{
    if (aptx_check_parity(channels, idx)) {
        int i;
        Channel *c;
        static const int map[] = { 1, 2, 0, 3 };
        Quantize *min = &channels[NB_CHANNELS-1].quantize[map[0]];
        for (c = &channels[NB_CHANNELS-1]; c >= channels; c--)
            for (i = 0; i < NB_SUBBANDS; i++)
                if (c->quantize[map[i]].error < min->error)
                    min = &c->quantize[map[i]];

        /* Forcing the desired parity is done by offsetting by 1 the quantized
         * sample from the subband featuring the smallest quantization error. */
        min->quantized_sample = min->quantized_sample_parity_change;
    }
}

static uint16_t aptx_pack_codeword(Channel *channel)
{
    int32_t parity = aptx_quantized_parity(channel);
    return (((channel->quantize[3].quantized_sample & 0x06) | parity) << 13)
         | (((channel->quantize[2].quantized_sample & 0x03)         ) << 11)
         | (((channel->quantize[1].quantized_sample & 0x0F)         ) <<  7)
         | (((channel->quantize[0].quantized_sample & 0x7F)         ) <<  0);
}

static uint32_t aptxhd_pack_codeword(Channel *channel)
{
    int32_t parity = aptx_quantized_parity(channel);
    return (((channel->quantize[3].quantized_sample & 0x01E) | parity) << 19)
         | (((channel->quantize[2].quantized_sample & 0x00F)         ) << 15)
         | (((channel->quantize[1].quantized_sample & 0x03F)         ) <<  9)
         | (((channel->quantize[0].quantized_sample & 0x1FF)         ) <<  0);
}

static void aptx_encode_samples(AptXContext *ctx,
                                int32_t samples[NB_CHANNELS][4],
                                uint8_t *output)
{
    int channel;
    for (channel = 0; channel < NB_CHANNELS; channel++)
        aptx_encode_channel(&ctx->channels[channel], samples[channel], ctx->hd);

    aptx_insert_sync(ctx->channels, &ctx->sync_idx);

    for (channel = 0; channel < NB_CHANNELS; channel++) {
        ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd);
        if (ctx->hd)
            AV_WB24(output + 3*channel,
                    aptxhd_pack_codeword(&ctx->channels[channel]));
        else
            AV_WB16(output + 2*channel,
                    aptx_pack_codeword(&ctx->channels[channel]));
    }
}

static int aptx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                             const AVFrame *frame, int *got_packet_ptr)
{
    AptXContext *s = avctx->priv_data;
    int pos, ipos, channel, sample, output_size, ret;

    if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
        return ret;

    output_size = s->block_size * frame->nb_samples/4;
    if ((ret = ff_get_encode_buffer(avctx, avpkt, output_size, 0)) < 0)
        return ret;

    for (pos = 0, ipos = 0; pos < output_size; pos += s->block_size, ipos += 4) {
        int32_t samples[NB_CHANNELS][4];

        for (channel = 0; channel < NB_CHANNELS; channel++)
            for (sample = 0; sample < 4; sample++)
                samples[channel][sample] = (int32_t)AV_RN32A(&frame->data[channel][4*(ipos+sample)]) >> 8;

        aptx_encode_samples(s, samples, avpkt->data + pos);
    }

    ff_af_queue_remove(&s->afq, frame->nb_samples, &avpkt->pts, &avpkt->duration);
    *got_packet_ptr = 1;
    return 0;
}

static av_cold int aptx_close(AVCodecContext *avctx)
{
    AptXContext *s = avctx->priv_data;
    ff_af_queue_close(&s->afq);
    return 0;
}

#if CONFIG_APTX_ENCODER
const AVCodec ff_aptx_encoder = {
    .name                  = "aptx",
    .long_name             = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"),
    .type                  = AVMEDIA_TYPE_AUDIO,
    .id                    = AV_CODEC_ID_APTX,
    .capabilities          = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SMALL_LAST_FRAME,
    .priv_data_size        = sizeof(AptXContext),
    .init                  = ff_aptx_init,
    .encode2               = aptx_encode_frame,
    .close                 = aptx_close,
    .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 },
    .supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0},
};
#endif

#if CONFIG_APTX_HD_ENCODER
const AVCodec ff_aptx_hd_encoder = {
    .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,
    .capabilities          = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SMALL_LAST_FRAME,
    .priv_data_size        = sizeof(AptXContext),
    .init                  = ff_aptx_init,
    .encode2               = aptx_encode_frame,
    .close                 = aptx_close,
    .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 },
    .supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0},
};
#endif