2 * Copyright (c) CMU 1993 Computer Science, Speech Group
3 * Chengxiang Lu and Alex Hauptmann
4 * Copyright (c) 2005 Steve Underwood <steveu at coppice.org>
5 * Copyright (c) 2009 Kenan Gillet
6 * Copyright (c) 2010 Martin Storsjo
8 * This file is part of Libav.
10 * Libav is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * Libav is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with Libav; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27 * G.722 ADPCM audio decoder
29 * This G.722 decoder is a bit-exact implementation of the ITU G.722
30 * specification for all three specified bitrates - 64000bps, 56000bps
31 * and 48000bps. It passes the ITU tests.
33 * @note For the 56000bps and 48000bps bitrates, the lowest 1 or 2 bits
34 * respectively of each byte are ignored.
41 static av_cold int g722_decode_init(AVCodecContext * avctx)
43 G722Context *c = avctx->priv_data;
45 if (avctx->channels != 1) {
46 av_log(avctx, AV_LOG_ERROR, "Only mono tracks are allowed.\n");
47 return AVERROR_INVALIDDATA;
49 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
51 switch (avctx->bits_per_coded_sample) {
57 av_log(avctx, AV_LOG_WARNING, "Unsupported bits_per_coded_sample [%d], "
59 avctx->bits_per_coded_sample);
61 avctx->bits_per_coded_sample = 8;
65 c->band[0].scale_factor = 8;
66 c->band[1].scale_factor = 2;
67 c->prev_samples_pos = 22;
72 static const int16_t low_inv_quant5[32] = {
73 -35, -35, -2919, -2195, -1765, -1458, -1219, -1023,
74 -858, -714, -587, -473, -370, -276, -190, -110,
75 2919, 2195, 1765, 1458, 1219, 1023, 858, 714,
76 587, 473, 370, 276, 190, 110, 35, -35
79 static const int16_t *low_inv_quants[3] = { ff_g722_low_inv_quant6,
81 ff_g722_low_inv_quant4 };
83 static int g722_decode_frame(AVCodecContext *avctx, void *data,
84 int *data_size, AVPacket *avpkt)
86 G722Context *c = avctx->priv_data;
87 int16_t *out_buf = data;
89 const int skip = 8 - avctx->bits_per_coded_sample;
90 const int16_t *quantizer_table = low_inv_quants[skip];
93 out_len = avpkt->size * 2 * av_get_bytes_per_sample(avctx->sample_fmt);
94 if (*data_size < out_len) {
95 av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
96 return AVERROR(EINVAL);
99 init_get_bits(&gb, avpkt->data, avpkt->size * 8);
101 for (j = 0; j < avpkt->size; j++) {
102 int ilow, ihigh, rlow, rhigh, dhigh;
105 ihigh = get_bits(&gb, 2);
106 ilow = get_bits(&gb, 6 - skip);
107 skip_bits(&gb, skip);
109 rlow = av_clip((c->band[0].scale_factor * quantizer_table[ilow] >> 10)
110 + c->band[0].s_predictor, -16384, 16383);
112 ff_g722_update_low_predictor(&c->band[0], ilow >> (2 - skip));
114 dhigh = c->band[1].scale_factor * ff_g722_high_inv_quant[ihigh] >> 10;
115 rhigh = av_clip(dhigh + c->band[1].s_predictor, -16384, 16383);
117 ff_g722_update_high_predictor(&c->band[1], dhigh, ihigh);
119 c->prev_samples[c->prev_samples_pos++] = rlow + rhigh;
120 c->prev_samples[c->prev_samples_pos++] = rlow - rhigh;
121 ff_g722_apply_qmf(c->prev_samples + c->prev_samples_pos - 24,
123 *out_buf++ = av_clip_int16(xout1 >> 12);
124 *out_buf++ = av_clip_int16(xout2 >> 12);
125 if (c->prev_samples_pos >= PREV_SAMPLES_BUF_SIZE) {
126 memmove(c->prev_samples, c->prev_samples + c->prev_samples_pos - 22,
127 22 * sizeof(c->prev_samples[0]));
128 c->prev_samples_pos = 22;
131 *data_size = out_len;
135 AVCodec ff_adpcm_g722_decoder = {
137 .type = AVMEDIA_TYPE_AUDIO,
138 .id = CODEC_ID_ADPCM_G722,
139 .priv_data_size = sizeof(G722Context),
140 .init = g722_decode_init,
141 .decode = g722_decode_frame,
142 .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),