GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/fastaudio.c Lines: 0 86 0.0 %
Date: 2021-04-14 23:45:22 Branches: 0 54 0.0 %

Line Branch Exec Source
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/*
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 * MOFLEX Fast Audio decoder
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 * Copyright (c) 2015-2016 Florian Nouwt
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 * Copyright (c) 2017 Adib Surani
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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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};