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authorPaul B Mahol <onemda@gmail.com>2022-09-07 13:58:53 +0200
committerPaul B Mahol <onemda@gmail.com>2022-09-12 11:34:27 +0200
commit88170070c4036530b79690c70e603fdddbb021c4 (patch)
tree83345cbae3ad4de3c6c463f25f559aa8b39bebb8 /libavcodec/bonk.c
parent5c19cb3f924c8afafcae08916a4167e36842adcd (diff)
downloadffmpeg-88170070c4036530b79690c70e603fdddbb021c4.tar.gz
avcodec: add bonk audio decoder
Diffstat (limited to 'libavcodec/bonk.c')
-rw-r--r--libavcodec/bonk.c433
1 files changed, 433 insertions, 0 deletions
diff --git a/libavcodec/bonk.c b/libavcodec/bonk.c
new file mode 100644
index 0000000000..f3d797d588
--- /dev/null
+++ b/libavcodec/bonk.c
@@ -0,0 +1,433 @@
+/*
+ * 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;
+ int i, bits;
+
+ if (max == 0)
+ return 0;
+
+ if (max >> 31)
+ return 32;
+
+ bits = 32 - ff_clz(max);
+
+ for (i = 0; i < bits - 1; i++)
+ if (get_bits1(&s->gb))
+ value += 1 << i;
+
+ if ((value | (1 << (bits - 1))) <= max)
+ if (get_bits1(&s->gb))
+ value += 1 << (bits - 1);
+
+ 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--) {
+ int 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
+ 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 },
+};