/* * Bonk audio decoder * * 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/internal.h" #include "libavutil/intreadwrite.h" #include "avcodec.h" #include "codec_internal.h" #include "decode.h" #define BITSTREAM_READER_LE #include "get_bits.h" #include "bytestream.h" typedef struct BitCount { uint8_t bit; unsigned count; } BitCount; typedef struct BonkContext { GetBitContext gb; int skip; uint8_t *bitstream; int64_t max_framesize; int bitstream_size; int bitstream_index; uint64_t nb_samples; int lossless; int mid_side; int n_taps; int down_sampling; int samples_per_packet; int state[2][2048], k[2048]; int *samples[2]; int *input_samples; uint8_t quant[2048]; BitCount *bits; } BonkContext; static av_cold int bonk_close(AVCodecContext *avctx) { BonkContext *s = avctx->priv_data; av_freep(&s->bitstream); av_freep(&s->input_samples); av_freep(&s->samples[0]); av_freep(&s->samples[1]); av_freep(&s->bits); s->bitstream_size = 0; return 0; } static av_cold int bonk_init(AVCodecContext *avctx) { BonkContext *s = avctx->priv_data; avctx->sample_fmt = AV_SAMPLE_FMT_S16P; if (avctx->extradata_size < 17) return AVERROR(EINVAL); if (avctx->extradata[0]) { av_log(avctx, AV_LOG_ERROR, "Unsupported version.\n"); return AVERROR_INVALIDDATA; } if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) return AVERROR_INVALIDDATA; s->nb_samples = AV_RL32(avctx->extradata + 1) / avctx->ch_layout.nb_channels; if (!s->nb_samples) s->nb_samples = UINT64_MAX; s->lossless = avctx->extradata[10] != 0; s->mid_side = avctx->extradata[11] != 0; s->n_taps = AV_RL16(avctx->extradata + 12); if (!s->n_taps || s->n_taps > 2048) return AVERROR(EINVAL); s->down_sampling = avctx->extradata[14]; if (!s->down_sampling) return AVERROR(EINVAL); s->samples_per_packet = AV_RL16(avctx->extradata + 15); if (!s->samples_per_packet) return AVERROR(EINVAL); s->max_framesize = s->samples_per_packet * avctx->ch_layout.nb_channels * s->down_sampling * 16LL; if (s->max_framesize > (INT32_MAX - AV_INPUT_BUFFER_PADDING_SIZE) / 8) return AVERROR_INVALIDDATA; s->bitstream = av_calloc(s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream)); if (!s->bitstream) return AVERROR(ENOMEM); s->input_samples = av_calloc(s->samples_per_packet, sizeof(*s->input_samples)); if (!s->input_samples) return AVERROR(ENOMEM); s->samples[0] = av_calloc(s->samples_per_packet * s->down_sampling, sizeof(*s->samples[0])); s->samples[1] = av_calloc(s->samples_per_packet * s->down_sampling, sizeof(*s->samples[0])); if (!s->samples[0] || !s->samples[1]) return AVERROR(ENOMEM); s->bits = av_calloc(s->max_framesize * 8, sizeof(*s->bits)); if (!s->bits) return AVERROR(ENOMEM); for (int i = 0; i < 512; i++) { s->quant[i] = sqrt(i + 1); } return 0; } static unsigned read_uint_max(BonkContext *s, uint32_t max) { unsigned value = 0; if (max == 0) return 0; if (max >> 31) return 32; for (unsigned i = 1; i <= max - value; i+=i) if (get_bits1(&s->gb)) value += i; return value; } static int intlist_read(BonkContext *s, int *buf, int entries, int base_2_part) { int i, low_bits = 0, x = 0, max_x; int n_zeros = 0, step = 256, dominant = 0; int pos = 0, level = 0; BitCount *bits = s->bits; memset(buf, 0, entries * sizeof(*buf)); if (base_2_part) { low_bits = get_bits(&s->gb, 4); if (low_bits) for (i = 0; i < entries; i++) buf[i] = get_bits(&s->gb, low_bits); } while (n_zeros < entries) { int steplet = step >> 8; if (get_bits_left(&s->gb) <= 0) return AVERROR_INVALIDDATA; if (!get_bits1(&s->gb)) { if (steplet < 0) break; if (steplet > 0) { bits[x ].bit = dominant; bits[x++].count = steplet; } if (!dominant) n_zeros += steplet; step += step / 8; } else if (steplet > 0) { int actual_run = read_uint_max(s, steplet - 1); if (actual_run < 0) break; if (actual_run > 0) { bits[x ].bit = dominant; bits[x++].count = actual_run; } bits[x ].bit = !dominant; bits[x++].count = 1; if (!dominant) n_zeros += actual_run; else n_zeros++; step -= step / 8; } if (step < 256) { if (step == 0) return AVERROR_INVALIDDATA; step = 65536 / step; dominant = !dominant; } } max_x = x; x = 0; n_zeros = 0; for (i = 0; n_zeros < entries; i++) { if (pos >= entries) { pos = 0; level += 1 << low_bits; } if (x >= max_x) return AVERROR_INVALIDDATA; if (buf[pos] >= level) { if (bits[x].bit) buf[pos] += 1 << low_bits; else n_zeros++; bits[x].count--; x += bits[x].count == 0; } pos++; } for (i = 0; i < entries; i++) { if (buf[i] && get_bits1(&s->gb)) { buf[i] = -buf[i]; } } return 0; } static inline int shift_down(int a, int b) { return (a >> b) + (a < 0); } static inline int shift(int a, int b) { return a + (1 << b - 1) >> b; } #define LATTICE_SHIFT 10 #define SAMPLE_SHIFT 4 #define SAMPLE_FACTOR (1 << SAMPLE_SHIFT) static int predictor_calc_error(int *k, int *state, int order, int error) { int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT); int *k_ptr = &(k[order-2]), *state_ptr = &(state[order-2]); for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--) { unsigned k_value = *k_ptr, state_value = *state_ptr; x -= shift_down(k_value * state_value, LATTICE_SHIFT); state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT); } // don't drift too far, to avoid overflows x = av_clip(x, -(SAMPLE_FACTOR << 16), SAMPLE_FACTOR << 16); state[0] = x; return x; } static void predictor_init_state(int *k, int *state, int order) { for (int i = order - 2; i >= 0; i--) { int x = state[i]; for (int j = 0, p = i + 1; p < order; j++, p++) { int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT); state[p] += shift_down(k[j] * x, LATTICE_SHIFT); x = tmp; } } } static int bonk_decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *pkt) { BonkContext *s = avctx->priv_data; GetBitContext *gb = &s->gb; const uint8_t *buf; int quant, n, buf_size, input_buf_size; int ret = AVERROR_INVALIDDATA; if ((!pkt->size && !s->bitstream_size) || s->nb_samples == 0) { *got_frame_ptr = 0; return pkt->size; } buf_size = FFMIN(pkt->size, s->max_framesize - s->bitstream_size); input_buf_size = buf_size; if (s->bitstream_index + s->bitstream_size + buf_size + AV_INPUT_BUFFER_PADDING_SIZE > s->max_framesize) { memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size); s->bitstream_index = 0; } if (pkt->data) memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size); buf = &s->bitstream[s->bitstream_index]; buf_size += s->bitstream_size; s->bitstream_size = buf_size; if (buf_size < s->max_framesize && pkt->data) { *got_frame_ptr = 0; return input_buf_size; } frame->nb_samples = FFMIN(s->samples_per_packet * s->down_sampling, s->nb_samples); if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; if ((ret = init_get_bits8(gb, buf, buf_size)) < 0) return ret; skip_bits(gb, s->skip); if ((ret = intlist_read(s, s->k, s->n_taps, 0)) < 0) return ret; for (int i = 0; i < s->n_taps; i++) s->k[i] *= s->quant[i]; quant = s->lossless ? 1 : get_bits(&s->gb, 16) * SAMPLE_FACTOR; for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) { const int samples_per_packet = s->samples_per_packet; const int down_sampling = s->down_sampling; const int offset = samples_per_packet * down_sampling - 1; int *state = s->state[ch]; int *sample = s->samples[ch]; predictor_init_state(s->k, state, s->n_taps); if ((ret = intlist_read(s, s->input_samples, samples_per_packet, 1)) < 0) return ret; for (int i = 0; i < samples_per_packet; i++) { for (int j = 0; j < s->down_sampling - 1; j++) { sample[0] = predictor_calc_error(s->k, state, s->n_taps, 0); sample++; } sample[0] = predictor_calc_error(s->k, state, s->n_taps, s->input_samples[i] * quant); sample++; } sample = s->samples[ch]; for (int i = 0; i < s->n_taps; i++) state[i] = sample[offset - i]; } if (s->mid_side && avctx->ch_layout.nb_channels == 2) { for (int i = 0; i < frame->nb_samples; i++) { s->samples[1][i] += shift(s->samples[0][i], 1); s->samples[0][i] -= s->samples[1][i]; } } if (!s->lossless) { for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) { int *samples = s->samples[ch]; for (int i = 0; i < frame->nb_samples; i++) samples[i] = shift(samples[i], 4); } } for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) { int16_t *osamples = (int16_t *)frame->extended_data[ch]; int *samples = s->samples[ch]; for (int i = 0; i < frame->nb_samples; i++) osamples[i] = av_clip_int16(samples[i]); } s->nb_samples -= frame->nb_samples; s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8); n = get_bits_count(gb) / 8; if (n > buf_size) { s->bitstream_size = 0; s->bitstream_index = 0; return AVERROR_INVALIDDATA; } *got_frame_ptr = 1; if (s->bitstream_size) { s->bitstream_index += n; s->bitstream_size -= n; return input_buf_size; } return n; } const FFCodec ff_bonk_decoder = { .p.name = "bonk", CODEC_LONG_NAME("Bonk audio"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_BONK, .priv_data_size = sizeof(BonkContext), .init = bonk_init, FF_CODEC_DECODE_CB(bonk_decode), .close = bonk_close, .p.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SUBFRAMES, .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE }, };