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https://github.com/FFmpeg/FFmpeg.git
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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
492 lines
14 KiB
C
492 lines
14 KiB
C
/*
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* OSQ audio decoder
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* Copyright (c) 2023 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/internal.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/mem.h"
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#include "avcodec.h"
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#include "codec_internal.h"
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#include "decode.h"
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#include "internal.h"
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#define BITSTREAM_READER_LE
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#include "get_bits.h"
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#include "unary.h"
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#define OFFSET 5
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typedef struct OSQChannel {
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unsigned prediction;
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unsigned coding_mode;
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unsigned residue_parameter;
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unsigned residue_bits;
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unsigned history[3];
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unsigned pos, count;
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double sum;
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int32_t prev;
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} OSQChannel;
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typedef struct OSQContext {
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GetBitContext gb;
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OSQChannel ch[2];
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uint8_t *bitstream;
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size_t max_framesize;
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size_t bitstream_size;
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int factor;
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int decorrelate;
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int frame_samples;
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uint64_t nb_samples;
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int32_t *decode_buffer[2];
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AVPacket *pkt;
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int pkt_offset;
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} OSQContext;
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static void osq_flush(AVCodecContext *avctx)
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{
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OSQContext *s = avctx->priv_data;
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s->bitstream_size = 0;
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s->pkt_offset = 0;
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}
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static av_cold int osq_close(AVCodecContext *avctx)
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{
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OSQContext *s = avctx->priv_data;
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av_freep(&s->bitstream);
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s->bitstream_size = 0;
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for (int ch = 0; ch < FF_ARRAY_ELEMS(s->decode_buffer); ch++)
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av_freep(&s->decode_buffer[ch]);
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return 0;
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}
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static av_cold int osq_init(AVCodecContext *avctx)
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{
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OSQContext *s = avctx->priv_data;
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if (avctx->extradata_size < 48)
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return AVERROR(EINVAL);
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if (avctx->extradata[0] != 1) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported version.\n");
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return AVERROR_INVALIDDATA;
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}
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avctx->sample_rate = AV_RL32(avctx->extradata + 4);
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if (avctx->sample_rate < 1)
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return AVERROR_INVALIDDATA;
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av_channel_layout_uninit(&avctx->ch_layout);
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avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
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avctx->ch_layout.nb_channels = avctx->extradata[3];
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if (avctx->ch_layout.nb_channels < 1)
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return AVERROR_INVALIDDATA;
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if (avctx->ch_layout.nb_channels > FF_ARRAY_ELEMS(s->decode_buffer))
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return AVERROR_INVALIDDATA;
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s->factor = 1;
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switch (avctx->extradata[2]) {
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case 8: avctx->sample_fmt = AV_SAMPLE_FMT_U8P; break;
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case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break;
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case 20:
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case 24: s->factor = 256;
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avctx->sample_fmt = AV_SAMPLE_FMT_S32P; break;
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default: return AVERROR_INVALIDDATA;
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}
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avctx->bits_per_raw_sample = avctx->extradata[2];
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s->nb_samples = AV_RL64(avctx->extradata + 16);
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s->frame_samples = AV_RL16(avctx->extradata + 8);
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s->max_framesize = (s->frame_samples * 16 + 1024) * avctx->ch_layout.nb_channels;
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s->bitstream = av_calloc(s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));
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if (!s->bitstream)
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return AVERROR(ENOMEM);
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for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
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s->decode_buffer[ch] = av_calloc(s->frame_samples + OFFSET,
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sizeof(*s->decode_buffer[ch]));
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if (!s->decode_buffer[ch])
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return AVERROR(ENOMEM);
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}
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s->pkt = avctx->internal->in_pkt;
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return 0;
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}
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static void reset_stats(OSQChannel *cb)
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{
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memset(cb->history, 0, sizeof(cb->history));
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cb->pos = cb->count = cb->sum = 0;
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}
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static void update_stats(OSQChannel *cb, int val)
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{
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cb->sum += FFABS(val) - cb->history[cb->pos];
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cb->history[cb->pos] = FFABS(val);
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cb->pos++;
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cb->count++;
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if (cb->pos >= FF_ARRAY_ELEMS(cb->history))
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cb->pos = 0;
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}
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static int update_residue_parameter(OSQChannel *cb)
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{
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double sum, x;
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int rice_k;
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sum = cb->sum;
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x = sum / cb->count;
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rice_k = av_ceil_log2(x);
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if (rice_k >= 30) {
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rice_k = floor(sum / 1.4426952 + 0.5);
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if (rice_k < 1)
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rice_k = 1;
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}
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return rice_k;
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}
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static uint32_t get_urice(GetBitContext *gb, int k)
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{
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uint32_t z, x, b;
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x = get_unary(gb, 1, 512);
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b = get_bits_long(gb, k);
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z = b | x << k;
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return z;
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}
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static int32_t get_srice(GetBitContext *gb, int x)
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{
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int32_t y = get_urice(gb, x);
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return get_bits1(gb) ? -y : y;
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}
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static int osq_channel_parameters(AVCodecContext *avctx, int ch)
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{
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OSQContext *s = avctx->priv_data;
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OSQChannel *cb = &s->ch[ch];
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GetBitContext *gb = &s->gb;
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cb->prev = 0;
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cb->prediction = get_urice(gb, 5);
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cb->coding_mode = get_urice(gb, 3);
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if (cb->prediction >= 15)
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return AVERROR_INVALIDDATA;
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if (cb->coding_mode > 0 && cb->coding_mode < 3) {
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cb->residue_parameter = get_urice(gb, 4);
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if (!cb->residue_parameter || cb->residue_parameter >= 31)
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return AVERROR_INVALIDDATA;
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} else if (cb->coding_mode == 3) {
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cb->residue_bits = get_urice(gb, 4);
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if (!cb->residue_bits || cb->residue_bits >= 31)
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return AVERROR_INVALIDDATA;
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} else if (cb->coding_mode) {
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return AVERROR_INVALIDDATA;
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}
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if (cb->coding_mode == 2)
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reset_stats(cb);
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return 0;
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}
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#define A (-1)
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#define B (-2)
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#define C (-3)
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#define D (-4)
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#define E (-5)
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#define P2 (((unsigned)dst[A] + dst[A]) - dst[B])
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#define P3 (((unsigned)dst[A] - dst[B]) * 3 + dst[C])
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static int do_decode(AVCodecContext *avctx, AVFrame *frame, int decorrelate, int downsample)
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{
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OSQContext *s = avctx->priv_data;
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const int nb_channels = avctx->ch_layout.nb_channels;
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const int nb_samples = frame->nb_samples;
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GetBitContext *gb = &s->gb;
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for (int n = 0; n < nb_samples; n++) {
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for (int ch = 0; ch < nb_channels; ch++) {
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OSQChannel *cb = &s->ch[ch];
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int32_t *dst = s->decode_buffer[ch] + OFFSET;
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int32_t p, prev = cb->prev;
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if (nb_channels == 2 && ch == 1 && decorrelate != s->decorrelate) {
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if (!decorrelate) {
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s->decode_buffer[1][OFFSET+A] += s->decode_buffer[0][OFFSET+B];
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s->decode_buffer[1][OFFSET+B] += s->decode_buffer[0][OFFSET+C];
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s->decode_buffer[1][OFFSET+C] += s->decode_buffer[0][OFFSET+D];
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s->decode_buffer[1][OFFSET+D] += s->decode_buffer[0][OFFSET+E];
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} else {
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s->decode_buffer[1][OFFSET+A] -= s->decode_buffer[0][OFFSET+B];
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s->decode_buffer[1][OFFSET+B] -= s->decode_buffer[0][OFFSET+C];
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s->decode_buffer[1][OFFSET+C] -= s->decode_buffer[0][OFFSET+D];
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s->decode_buffer[1][OFFSET+D] -= s->decode_buffer[0][OFFSET+E];
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}
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s->decorrelate = decorrelate;
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}
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if (!cb->coding_mode) {
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dst[n] = 0;
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} else if (cb->coding_mode == 3) {
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dst[n] = get_sbits_long(gb, cb->residue_bits);
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} else {
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dst[n] = get_srice(gb, cb->residue_parameter);
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}
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if (get_bits_left(gb) < 0) {
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av_log(avctx, AV_LOG_ERROR, "overread!\n");
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return AVERROR_INVALIDDATA;
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}
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p = prev / 2;
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prev = dst[n];
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switch (cb->prediction) {
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case 0:
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break;
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case 1:
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dst[n] += (unsigned)dst[A];
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break;
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case 2:
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dst[n] += (unsigned)dst[A] + p;
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break;
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case 3:
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dst[n] += P2;
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break;
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case 4:
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dst[n] += P2 + p;
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break;
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case 5:
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dst[n] += P3;
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break;
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case 6:
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dst[n] += P3 + p;
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break;
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case 7:
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dst[n] += (int)(P2 + P3) / 2 + (unsigned)p;
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break;
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case 8:
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dst[n] += (int)(P2 + P3) / 2;
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break;
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case 9:
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dst[n] += (int)(P2 * 2 + P3) / 3 + (unsigned)p;
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break;
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case 10:
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dst[n] += (int)(P2 + P3 * 2) / 3 + (unsigned)p;
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break;
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case 11:
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dst[n] += (int)((unsigned)dst[A] + dst[B]) / 2;
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break;
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case 12:
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dst[n] += (unsigned)dst[B];
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break;
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case 13:
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dst[n] += (int)(unsigned)(dst[D] + dst[B]) / 2;
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break;
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case 14:
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dst[n] += (int)((unsigned)P2 + dst[A]) / 2 + (unsigned)p;
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break;
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default:
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return AVERROR_INVALIDDATA;
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}
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cb->prev = prev;
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if (downsample)
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dst[n] *= 256;
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dst[E] = dst[D];
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dst[D] = dst[C];
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dst[C] = dst[B];
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dst[B] = dst[A];
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dst[A] = dst[n];
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if (cb->coding_mode == 2) {
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update_stats(cb, dst[n]);
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cb->residue_parameter = update_residue_parameter(cb);
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}
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if (nb_channels == 2 && ch == 1) {
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if (decorrelate)
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dst[n] += s->decode_buffer[0][OFFSET+n];
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}
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if (downsample)
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dst[A] /= 256;
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}
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}
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return 0;
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}
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static int osq_decode_block(AVCodecContext *avctx, AVFrame *frame)
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{
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const int nb_channels = avctx->ch_layout.nb_channels;
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const int nb_samples = frame->nb_samples;
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OSQContext *s = avctx->priv_data;
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const int factor = s->factor;
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int ret, decorrelate, downsample;
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GetBitContext *gb = &s->gb;
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skip_bits1(gb);
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decorrelate = get_bits1(gb);
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downsample = get_bits1(gb);
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for (int ch = 0; ch < nb_channels; ch++) {
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if ((ret = osq_channel_parameters(avctx, ch)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "invalid channel parameters\n");
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return ret;
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}
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}
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if ((ret = do_decode(avctx, frame, decorrelate, downsample)) < 0)
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return ret;
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align_get_bits(gb);
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switch (avctx->sample_fmt) {
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case AV_SAMPLE_FMT_U8P:
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for (int ch = 0; ch < nb_channels; ch++) {
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uint8_t *dst = (uint8_t *)frame->extended_data[ch];
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int32_t *src = s->decode_buffer[ch] + OFFSET;
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for (int n = 0; n < nb_samples; n++)
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dst[n] = av_clip_uint8(src[n] + 0x80);
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}
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break;
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case AV_SAMPLE_FMT_S16P:
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for (int ch = 0; ch < nb_channels; ch++) {
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int16_t *dst = (int16_t *)frame->extended_data[ch];
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int32_t *src = s->decode_buffer[ch] + OFFSET;
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for (int n = 0; n < nb_samples; n++)
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dst[n] = (int16_t)src[n];
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}
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break;
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case AV_SAMPLE_FMT_S32P:
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for (int ch = 0; ch < nb_channels; ch++) {
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int32_t *dst = (int32_t *)frame->extended_data[ch];
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int32_t *src = s->decode_buffer[ch] + OFFSET;
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for (int n = 0; n < nb_samples; n++)
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dst[n] = src[n] * factor;
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}
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break;
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default:
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return AVERROR_BUG;
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}
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return 0;
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}
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static int osq_receive_frame(AVCodecContext *avctx, AVFrame *frame)
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{
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OSQContext *s = avctx->priv_data;
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GetBitContext *gb = &s->gb;
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int ret, n;
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while (s->bitstream_size < s->max_framesize) {
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int size;
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if (!s->pkt->data) {
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ret = ff_decode_get_packet(avctx, s->pkt);
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if (ret == AVERROR_EOF && s->bitstream_size > 0)
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break;
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if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
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return ret;
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if (ret < 0)
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goto fail;
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}
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size = FFMIN(s->pkt->size - s->pkt_offset, s->max_framesize - s->bitstream_size);
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memcpy(s->bitstream + s->bitstream_size, s->pkt->data + s->pkt_offset, size);
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s->bitstream_size += size;
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s->pkt_offset += size;
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if (s->pkt_offset == s->pkt->size) {
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av_packet_unref(s->pkt);
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s->pkt_offset = 0;
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}
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}
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frame->nb_samples = FFMIN(s->frame_samples, s->nb_samples);
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if (frame->nb_samples <= 0)
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return AVERROR_EOF;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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goto fail;
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if ((ret = init_get_bits8(gb, s->bitstream, s->bitstream_size)) < 0)
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goto fail;
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if ((ret = osq_decode_block(avctx, frame)) < 0)
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goto fail;
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s->nb_samples -= frame->nb_samples;
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n = get_bits_count(gb) / 8;
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if (n > s->bitstream_size) {
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ret = AVERROR_INVALIDDATA;
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goto fail;
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}
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memmove(s->bitstream, &s->bitstream[n], s->bitstream_size - n);
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s->bitstream_size -= n;
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return 0;
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fail:
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s->bitstream_size = 0;
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s->pkt_offset = 0;
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av_packet_unref(s->pkt);
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return ret;
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}
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const FFCodec ff_osq_decoder = {
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.p.name = "osq",
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CODEC_LONG_NAME("OSQ (Original Sound Quality)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_OSQ,
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.priv_data_size = sizeof(OSQContext),
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.init = osq_init,
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FF_CODEC_RECEIVE_FRAME_CB(osq_receive_frame),
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.close = osq_close,
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.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
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.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_NONE },
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.flush = osq_flush,
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};
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