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/* |
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* MOFLEX Fast Audio decoder |
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* Copyright (c) 2020 Paul B Mahol |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "libavutil/intreadwrite.h" |
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#include "avcodec.h" |
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#include "bytestream.h" |
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#include "internal.h" |
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#include "mathops.h" |
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typedef struct ChannelItems { |
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float f[8]; |
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float last; |
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} ChannelItems; |
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typedef struct FastAudioContext { |
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float table[8][64]; |
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ChannelItems *ch; |
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} FastAudioContext; |
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static av_cold int fastaudio_init(AVCodecContext *avctx) |
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{ |
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FastAudioContext *s = avctx->priv_data; |
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; |
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for (int i = 0; i < 8; i++) |
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s->table[0][i] = (i - 159.5f) / 160.f; |
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for (int i = 0; i < 11; i++) |
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s->table[0][i + 8] = (i - 37.5f) / 40.f; |
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for (int i = 0; i < 27; i++) |
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s->table[0][i + 8 + 11] = (i - 13.f) / 20.f; |
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for (int i = 0; i < 11; i++) |
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s->table[0][i + 8 + 11 + 27] = (i + 27.5f) / 40.f; |
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for (int i = 0; i < 7; i++) |
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s->table[0][i + 8 + 11 + 27 + 11] = (i + 152.5f) / 160.f; |
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memcpy(s->table[1], s->table[0], sizeof(s->table[0])); |
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for (int i = 0; i < 7; i++) |
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s->table[2][i] = (i - 33.5f) / 40.f; |
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for (int i = 0; i < 25; i++) |
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s->table[2][i + 7] = (i - 13.f) / 20.f; |
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for (int i = 0; i < 32; i++) |
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s->table[3][i] = -s->table[2][31 - i]; |
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for (int i = 0; i < 16; i++) |
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s->table[4][i] = i * 0.22f / 3.f - 0.6f; |
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for (int i = 0; i < 16; i++) |
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s->table[5][i] = i * 0.20f / 3.f - 0.3f; |
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for (int i = 0; i < 8; i++) |
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s->table[6][i] = i * 0.36f / 3.f - 0.4f; |
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for (int i = 0; i < 8; i++) |
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s->table[7][i] = i * 0.34f / 3.f - 0.2f; |
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s->ch = av_calloc(avctx->channels, sizeof(*s->ch)); |
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if (!s->ch) |
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return AVERROR(ENOMEM); |
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return 0; |
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} |
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static int read_bits(int bits, int *ppos, unsigned *src) |
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{ |
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int r, pos; |
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pos = *ppos; |
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pos += bits; |
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r = src[(pos - 1) / 32] >> ((-pos) & 31); |
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*ppos = pos; |
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return r & ((1 << bits) - 1); |
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} |
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static const uint8_t bits[8] = { 6, 6, 5, 5, 4, 0, 3, 3, }; |
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static void set_sample(int i, int j, int v, float *result, int *pads, float value) |
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{ |
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result[i * 64 + pads[i] + j * 3] = value * (2 * v - 7); |
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} |
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static int fastaudio_decode(AVCodecContext *avctx, void *data, |
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int *got_frame, AVPacket *pkt) |
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{ |
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FastAudioContext *s = avctx->priv_data; |
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GetByteContext gb; |
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AVFrame *frame = data; |
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int subframes; |
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int ret; |
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subframes = pkt->size / (40 * avctx->channels); |
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frame->nb_samples = subframes * 256; |
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
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return ret; |
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bytestream2_init(&gb, pkt->data, pkt->size); |
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for (int subframe = 0; subframe < subframes; subframe++) { |
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for (int channel = 0; channel < avctx->channels; channel++) { |
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ChannelItems *ch = &s->ch[channel]; |
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float result[256] = { 0 }; |
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unsigned src[10]; |
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int inds[4], pads[4]; |
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float m[8]; |
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int pos = 0; |
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for (int i = 0; i < 10; i++) |
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src[i] = bytestream2_get_le32(&gb); |
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for (int i = 0; i < 8; i++) |
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m[7 - i] = s->table[i][read_bits(bits[i], &pos, src)]; |
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for (int i = 0; i < 4; i++) |
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inds[3 - i] = read_bits(6, &pos, src); |
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for (int i = 0; i < 4; i++) |
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pads[3 - i] = read_bits(2, &pos, src); |
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for (int i = 0, index5 = 0; i < 4; i++) { |
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float value = av_int2float((inds[i] + 1) << 20) * powf(2.f, 116.f); |
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for (int j = 0, tmp = 0; j < 21; j++) { |
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set_sample(i, j, j == 20 ? tmp / 2 : read_bits(3, &pos, src), result, pads, value); |
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if (j % 10 == 9) |
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tmp = 4 * tmp + read_bits(2, &pos, src); |
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if (j == 20) |
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index5 = FFMIN(2 * index5 + tmp % 2, 63); |
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} |
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m[2] = s->table[5][index5]; |
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} |
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for (int i = 0; i < 256; i++) { |
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float x = result[i]; |
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for (int j = 0; j < 8; j++) { |
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x -= m[j] * ch->f[j]; |
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ch->f[j] += m[j] * x; |
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} |
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memmove(&ch->f[0], &ch->f[1], sizeof(float) * 7); |
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ch->f[7] = x; |
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ch->last = x + ch->last * 0.86f; |
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result[i] = ch->last * 2.f; |
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} |
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memcpy(frame->extended_data[channel] + 1024 * subframe, result, 256 * sizeof(float)); |
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} |
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} |
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*got_frame = 1; |
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return pkt->size; |
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} |
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static av_cold int fastaudio_close(AVCodecContext *avctx) |
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{ |
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FastAudioContext *s = avctx->priv_data; |
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av_freep(&s->ch); |
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return 0; |
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} |
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AVCodec ff_fastaudio_decoder = { |
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.name = "fastaudio", |
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.long_name = NULL_IF_CONFIG_SMALL("MobiClip FastAudio"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_FASTAUDIO, |
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.priv_data_size = sizeof(FastAudioContext), |
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.init = fastaudio_init, |
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.decode = fastaudio_decode, |
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.close = fastaudio_close, |
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.capabilities = AV_CODEC_CAP_DR1, |
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, |
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AV_SAMPLE_FMT_NONE }, |
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}; |