Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 672 insertions, 0 deletions

diff --git a/media/ffvpx/libavcodec/flacdec.c b/media/ffvpx/libavcodec/flacdec.c
new file mode 100644
index 0000000000..5b8547a98f
--- /dev/null
+++ b/media/ffvpx/libavcodec/flacdec.c
@@ -0,0 +1,672 @@
+/*
+ * FLAC (Free Lossless Audio Codec) decoder
+ * Copyright (c) 2003 Alex Beregszaszi
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * FLAC (Free Lossless Audio Codec) decoder
+ * @author Alex Beregszaszi
+ * @see http://flac.sourceforge.net/
+ *
+ * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
+ * through, starting from the initial 'fLaC' signature; or by passing the
+ * 34-byte streaminfo structure through avctx->extradata[_size] followed
+ * by data starting with the 0xFFF8 marker.
+ */
+
+#include <limits.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/crc.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "codec_internal.h"
+#include "get_bits.h"
+#include "bytestream.h"
+#include "golomb.h"
+#include "flac.h"
+#include "flacdata.h"
+#include "flacdsp.h"
+#include "flac_parse.h"
+#include "thread.h"
+#include "unary.h"
+
+
+typedef struct FLACContext {
+    AVClass *class;
+    FLACStreaminfo stream_info;
+
+    AVCodecContext *avctx;                  ///< parent AVCodecContext
+    GetBitContext gb;                       ///< GetBitContext initialized to start at the current frame
+
+    int blocksize;                          ///< number of samples in the current frame
+    int sample_shift;                       ///< shift required to make output samples 16-bit or 32-bit
+    int ch_mode;                            ///< channel decorrelation type in the current frame
+    int got_streaminfo;                     ///< indicates if the STREAMINFO has been read
+
+    int32_t *decoded[FLAC_MAX_CHANNELS];    ///< decoded samples
+    uint8_t *decoded_buffer;
+    unsigned int decoded_buffer_size;
+    int buggy_lpc;                          ///< use workaround for old lavc encoded files
+
+    FLACDSPContext dsp;
+} FLACContext;
+
+static int allocate_buffers(FLACContext *s);
+
+static void flac_set_bps(FLACContext *s)
+{
+    enum AVSampleFormat req = s->avctx->request_sample_fmt;
+    int need32 = s->stream_info.bps > 16;
+    int want32 = av_get_bytes_per_sample(req) > 2;
+    int planar = av_sample_fmt_is_planar(req);
+
+    if (need32 || want32) {
+        if (planar)
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+        else
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+        s->sample_shift = 32 - s->stream_info.bps;
+    } else {
+        if (planar)
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+        else
+            s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+        s->sample_shift = 16 - s->stream_info.bps;
+    }
+}
+
+static av_cold int flac_decode_init(AVCodecContext *avctx)
+{
+    uint8_t *streaminfo;
+    int ret;
+    FLACContext *s = avctx->priv_data;
+    s->avctx = avctx;
+
+    /* for now, the raw FLAC header is allowed to be passed to the decoder as
+       frame data instead of extradata. */
+    if (!avctx->extradata)
+        return 0;
+
+    if (!ff_flac_is_extradata_valid(avctx, &streaminfo))
+        return AVERROR_INVALIDDATA;
+
+    /* initialize based on the demuxer-supplied streamdata header */
+    ret = ff_flac_parse_streaminfo(avctx, &s->stream_info, streaminfo);
+    if (ret < 0)
+        return ret;
+    ret = allocate_buffers(s);
+    if (ret < 0)
+        return ret;
+    flac_set_bps(s);
+    ff_flacdsp_init(&s->dsp, avctx->sample_fmt,
+                    s->stream_info.channels);
+    s->got_streaminfo = 1;
+
+    return 0;
+}
+
+static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
+{
+    av_log(avctx, AV_LOG_DEBUG, "  Max Blocksize: %d\n", s->max_blocksize);
+    av_log(avctx, AV_LOG_DEBUG, "  Max Framesize: %d\n", s->max_framesize);
+    av_log(avctx, AV_LOG_DEBUG, "  Samplerate: %d\n", s->samplerate);
+    av_log(avctx, AV_LOG_DEBUG, "  Channels: %d\n", s->channels);
+    av_log(avctx, AV_LOG_DEBUG, "  Bits: %d\n", s->bps);
+}
+
+static int allocate_buffers(FLACContext *s)
+{
+    int buf_size;
+    int ret;
+
+    av_assert0(s->stream_info.max_blocksize);
+
+    buf_size = av_samples_get_buffer_size(NULL, s->stream_info.channels,
+                                          s->stream_info.max_blocksize,
+                                          AV_SAMPLE_FMT_S32P, 0);
+    if (buf_size < 0)
+        return buf_size;
+
+    av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size);
+    if (!s->decoded_buffer)
+        return AVERROR(ENOMEM);
+
+    ret = av_samples_fill_arrays((uint8_t **)s->decoded, NULL,
+                                 s->decoded_buffer,
+                                 s->stream_info.channels,
+                                 s->stream_info.max_blocksize,
+                                 AV_SAMPLE_FMT_S32P, 0);
+    return ret < 0 ? ret : 0;
+}
+
+/**
+ * Parse the STREAMINFO from an inline header.
+ * @param s the flac decoding context
+ * @param buf input buffer, starting with the "fLaC" marker
+ * @param buf_size buffer size
+ * @return non-zero if metadata is invalid
+ */
+static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
+{
+    int metadata_type, metadata_size, ret;
+
+    if (buf_size < FLAC_STREAMINFO_SIZE+8) {
+        /* need more data */
+        return 0;
+    }
+    flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size);
+    if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO ||
+        metadata_size != FLAC_STREAMINFO_SIZE) {
+        return AVERROR_INVALIDDATA;
+    }
+    ret = ff_flac_parse_streaminfo(s->avctx, &s->stream_info, &buf[8]);
+    if (ret < 0)
+        return ret;
+    ret = allocate_buffers(s);
+    if (ret < 0)
+        return ret;
+    flac_set_bps(s);
+    ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt,
+                    s->stream_info.channels);
+    s->got_streaminfo = 1;
+
+    return 0;
+}
+
+/**
+ * Determine the size of an inline header.
+ * @param buf input buffer, starting with the "fLaC" marker
+ * @param buf_size buffer size
+ * @return number of bytes in the header, or 0 if more data is needed
+ */
+static int get_metadata_size(const uint8_t *buf, int buf_size)
+{
+    int metadata_last, metadata_size;
+    const uint8_t *buf_end = buf + buf_size;
+
+    buf += 4;
+    do {
+        if (buf_end - buf < 4)
+            return AVERROR_INVALIDDATA;
+        flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
+        buf += 4;
+        if (buf_end - buf < metadata_size) {
+            /* need more data in order to read the complete header */
+            return AVERROR_INVALIDDATA;
+        }
+        buf += metadata_size;
+    } while (!metadata_last);
+
+    return buf_size - (buf_end - buf);
+}
+
+static int decode_residuals(FLACContext *s, int32_t *decoded, int pred_order)
+{
+    GetBitContext gb = s->gb;
+    int i, tmp, partition, method_type, rice_order;
+    int rice_bits, rice_esc;
+    int samples;
+
+    method_type = get_bits(&gb, 2);
+    rice_order  = get_bits(&gb, 4);
+
+    samples   = s->blocksize >> rice_order;
+    rice_bits = 4 + method_type;
+    rice_esc  = (1 << rice_bits) - 1;
+
+    decoded += pred_order;
+    i        = pred_order;
+
+    if (method_type > 1) {
+        av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n",
+               method_type);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (samples << rice_order != s->blocksize) {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid rice order: %i blocksize %i\n",
+               rice_order, s->blocksize);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (pred_order > samples) {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n",
+               pred_order, samples);
+        return AVERROR_INVALIDDATA;
+    }
+
+    for (partition = 0; partition < (1 << rice_order); partition++) {
+        tmp = get_bits(&gb, rice_bits);
+        if (tmp == rice_esc) {
+            tmp = get_bits(&gb, 5);
+            for (; i < samples; i++)
+                *decoded++ = get_sbits_long(&gb, tmp);
+        } else {
+            int real_limit = (tmp > 1) ? (INT_MAX >> (tmp - 1)) + 2 : INT_MAX;
+            for (; i < samples; i++) {
+                int v = get_sr_golomb_flac(&gb, tmp, real_limit, 1);
+                if (v == 0x80000000){
+                    av_log(s->avctx, AV_LOG_ERROR, "invalid residual\n");
+                    return AVERROR_INVALIDDATA;
+                }
+
+                *decoded++ = v;
+            }
+        }
+        i= 0;
+    }
+
+    s->gb = gb;
+
+    return 0;
+}
+
+static int decode_subframe_fixed(FLACContext *s, int32_t *decoded,
+                                 int pred_order, int bps)
+{
+    const int blocksize = s->blocksize;
+    unsigned av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d);
+    int i;
+    int ret;
+
+    /* warm up samples */
+    for (i = 0; i < pred_order; i++) {
+        decoded[i] = get_sbits_long(&s->gb, bps);
+    }
+
+    if ((ret = decode_residuals(s, decoded, pred_order)) < 0)
+        return ret;
+
+    if (pred_order > 0)
+        a = decoded[pred_order-1];
+    if (pred_order > 1)
+        b = a - decoded[pred_order-2];
+    if (pred_order > 2)
+        c = b - decoded[pred_order-2] + decoded[pred_order-3];
+    if (pred_order > 3)
+        d = c - decoded[pred_order-2] + 2U*decoded[pred_order-3] - decoded[pred_order-4];
+
+    switch (pred_order) {
+    case 0:
+        break;
+    case 1:
+        for (i = pred_order; i < blocksize; i++)
+            decoded[i] = a += decoded[i];
+        break;
+    case 2:
+        for (i = pred_order; i < blocksize; i++)
+            decoded[i] = a += b += decoded[i];
+        break;
+    case 3:
+        for (i = pred_order; i < blocksize; i++)
+            decoded[i] = a += b += c += decoded[i];
+        break;
+    case 4:
+        for (i = pred_order; i < blocksize; i++)
+            decoded[i] = a += b += c += d += decoded[i];
+        break;
+    default:
+        av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
+        return AVERROR_INVALIDDATA;
+    }
+
+    return 0;
+}
+
+static void lpc_analyze_remodulate(SUINT32 *decoded, const int coeffs[32],
+                                   int order, int qlevel, int len, int bps)
+{
+    int i, j;
+    int ebps = 1 << (bps-1);
+    unsigned sigma = 0;
+
+    for (i = order; i < len; i++)
+        sigma |= decoded[i] + ebps;
+
+    if (sigma < 2*ebps)
+        return;
+
+    for (i = len - 1; i >= order; i--) {
+        int64_t p = 0;
+        for (j = 0; j < order; j++)
+            p += coeffs[j] * (int64_t)(int32_t)decoded[i-order+j];
+        decoded[i] -= p >> qlevel;
+    }
+    for (i = order; i < len; i++, decoded++) {
+        int32_t p = 0;
+        for (j = 0; j < order; j++)
+            p += coeffs[j] * (uint32_t)decoded[j];
+        decoded[j] += p >> qlevel;
+    }
+}
+
+static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order,
+                               int bps)
+{
+    int i, ret;
+    int coeff_prec, qlevel;
+    int coeffs[32];
+
+    /* warm up samples */
+    for (i = 0; i < pred_order; i++) {
+        decoded[i] = get_sbits_long(&s->gb, bps);
+    }
+
+    coeff_prec = get_bits(&s->gb, 4) + 1;
+    if (coeff_prec == 16) {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n");
+        return AVERROR_INVALIDDATA;
+    }
+    qlevel = get_sbits(&s->gb, 5);
+    if (qlevel < 0) {
+        av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n",
+               qlevel);
+        return AVERROR_INVALIDDATA;
+    }
+
+    for (i = 0; i < pred_order; i++) {
+        coeffs[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec);
+    }
+
+    if ((ret = decode_residuals(s, decoded, pred_order)) < 0)
+        return ret;
+
+    if (   (    s->buggy_lpc && s->stream_info.bps <= 16)
+        || (   !s->buggy_lpc && bps <= 16
+            && bps + coeff_prec + av_log2(pred_order) <= 32)) {
+        s->dsp.lpc16(decoded, coeffs, pred_order, qlevel, s->blocksize);
+    } else {
+        s->dsp.lpc32(decoded, coeffs, pred_order, qlevel, s->blocksize);
+        if (s->stream_info.bps <= 16)
+            lpc_analyze_remodulate(decoded, coeffs, pred_order, qlevel, s->blocksize, bps);
+    }
+
+    return 0;
+}
+
+static inline int decode_subframe(FLACContext *s, int channel)
+{
+    int32_t *decoded = s->decoded[channel];
+    int type, wasted = 0;
+    int bps = s->stream_info.bps;
+    int i, tmp, ret;
+
+    if (channel == 0) {
+        if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
+            bps++;
+    } else {
+        if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE)
+            bps++;
+    }
+
+    if (get_bits1(&s->gb)) {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
+        return AVERROR_INVALIDDATA;
+    }
+    type = get_bits(&s->gb, 6);
+
+    if (get_bits1(&s->gb)) {
+        int left = get_bits_left(&s->gb);
+        if ( left <= 0 ||
+            (left < bps && !show_bits_long(&s->gb, left)) ||
+                           !show_bits_long(&s->gb, bps)) {
+            av_log(s->avctx, AV_LOG_ERROR,
+                   "Invalid number of wasted bits > available bits (%d) - left=%d\n",
+                   bps, left);
+            return AVERROR_INVALIDDATA;
+        }
+        wasted = 1 + get_unary(&s->gb, 1, get_bits_left(&s->gb));
+        bps -= wasted;
+    }
+    if (bps > 32) {
+        avpriv_report_missing_feature(s->avctx, "Decorrelated bit depth > 32");
+        return AVERROR_PATCHWELCOME;
+    }
+
+//FIXME use av_log2 for types
+    if (type == 0) {
+        tmp = get_sbits_long(&s->gb, bps);
+        for (i = 0; i < s->blocksize; i++)
+            decoded[i] = tmp;
+    } else if (type == 1) {
+        for (i = 0; i < s->blocksize; i++)
+            decoded[i] = get_sbits_long(&s->gb, bps);
+    } else if ((type >= 8) && (type <= 12)) {
+        if ((ret = decode_subframe_fixed(s, decoded, type & ~0x8, bps)) < 0)
+            return ret;
+    } else if (type >= 32) {
+        if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0)
+            return ret;
+    } else {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (wasted && wasted < 32) {
+        int i;
+        for (i = 0; i < s->blocksize; i++)
+            decoded[i] = (unsigned)decoded[i] << wasted;
+    }
+
+    return 0;
+}
+
+static int decode_frame(FLACContext *s)
+{
+    int i, ret;
+    GetBitContext *gb = &s->gb;
+    FLACFrameInfo fi;
+
+    if ((ret = ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) < 0) {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n");
+        return ret;
+    }
+
+    if (   s->stream_info.channels
+        && fi.channels != s->stream_info.channels
+        && s->got_streaminfo) {
+        s->stream_info.channels = fi.channels;
+        ff_flac_set_channel_layout(s->avctx, fi.channels);
+        ret = allocate_buffers(s);
+        if (ret < 0)
+            return ret;
+    }
+    s->stream_info.channels = fi.channels;
+    ff_flac_set_channel_layout(s->avctx, fi.channels);
+    s->ch_mode = fi.ch_mode;
+
+    if (!s->stream_info.bps && !fi.bps) {
+        av_log(s->avctx, AV_LOG_ERROR, "bps not found in STREAMINFO or frame header\n");
+        return AVERROR_INVALIDDATA;
+    }
+    if (!fi.bps) {
+        fi.bps = s->stream_info.bps;
+    } else if (s->stream_info.bps && fi.bps != s->stream_info.bps) {
+        av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
+                                       "supported\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (!s->stream_info.bps) {
+        s->stream_info.bps = s->avctx->bits_per_raw_sample = fi.bps;
+        flac_set_bps(s);
+    }
+
+    if (!s->stream_info.max_blocksize)
+        s->stream_info.max_blocksize = FLAC_MAX_BLOCKSIZE;
+    if (fi.blocksize > s->stream_info.max_blocksize) {
+        av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize,
+               s->stream_info.max_blocksize);
+        return AVERROR_INVALIDDATA;
+    }
+    s->blocksize = fi.blocksize;
+
+    if (!s->stream_info.samplerate && !fi.samplerate) {
+        av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO"
+                                        " or frame header\n");
+        return AVERROR_INVALIDDATA;
+    }
+    if (fi.samplerate == 0)
+        fi.samplerate = s->stream_info.samplerate;
+    s->stream_info.samplerate = s->avctx->sample_rate = fi.samplerate;
+
+    if (!s->got_streaminfo) {
+        ret = allocate_buffers(s);
+        if (ret < 0)
+            return ret;
+        s->got_streaminfo = 1;
+        dump_headers(s->avctx, &s->stream_info);
+    }
+    ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt,
+                    s->stream_info.channels);
+
+//    dump_headers(s->avctx, &s->stream_info);
+
+    /* subframes */
+    for (i = 0; i < s->stream_info.channels; i++) {
+        if ((ret = decode_subframe(s, i)) < 0)
+            return ret;
+    }
+
+    align_get_bits(gb);
+
+    /* frame footer */
+    skip_bits(gb, 16); /* data crc */
+
+    return 0;
+}
+
+static int flac_decode_frame(AVCodecContext *avctx, AVFrame *frame,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    FLACContext *s = avctx->priv_data;
+    int bytes_read = 0;
+    int ret;
+
+    *got_frame_ptr = 0;
+
+    if (buf_size > 5 && !memcmp(buf, "\177FLAC", 5)) {
+        av_log(s->avctx, AV_LOG_DEBUG, "skipping flac header packet 1\n");
+        return buf_size;
+    }
+
+    if (buf_size > 0 && (*buf & 0x7F) == FLAC_METADATA_TYPE_VORBIS_COMMENT) {
+        av_log(s->avctx, AV_LOG_DEBUG, "skipping vorbis comment\n");
+        return buf_size;
+    }
+
+    /* check that there is at least the smallest decodable amount of data.
+       this amount corresponds to the smallest valid FLAC frame possible.
+       FF F8 69 02 00 00 9A 00 00 34 */
+    if (buf_size < FLAC_MIN_FRAME_SIZE)
+        return buf_size;
+
+    /* check for inline header */
+    if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
+        if (!s->got_streaminfo && (ret = parse_streaminfo(s, buf, buf_size))) {
+            av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
+            return ret;
+        }
+        return get_metadata_size(buf, buf_size);
+    }
+
+    /* decode frame */
+    if ((ret = init_get_bits8(&s->gb, buf, buf_size)) < 0)
+        return ret;
+    if ((ret = decode_frame(s)) < 0) {
+        av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
+        return ret;
+    }
+    bytes_read = get_bits_count(&s->gb)/8;
+
+    if ((s->avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_COMPLIANT)) &&
+        av_crc(av_crc_get_table(AV_CRC_16_ANSI),
+               0, buf, bytes_read)) {
+        av_log(s->avctx, AV_LOG_ERROR, "CRC error at PTS %"PRId64"\n", avpkt->pts);
+        if (s->avctx->err_recognition & AV_EF_EXPLODE)
+            return AVERROR_INVALIDDATA;
+    }
+
+    /* get output buffer */
+    frame->nb_samples = s->blocksize;
+    if ((ret = ff_thread_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+
+    s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded,
+                                   s->stream_info.channels,
+                                   s->blocksize, s->sample_shift);
+
+    if (bytes_read > buf_size) {
+        av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
+        return AVERROR_INVALIDDATA;
+    }
+    if (bytes_read < buf_size) {
+        av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n",
+               buf_size - bytes_read, buf_size);
+    }
+
+    *got_frame_ptr = 1;
+
+    return bytes_read;
+}
+
+static av_cold int flac_decode_close(AVCodecContext *avctx)
+{
+    FLACContext *s = avctx->priv_data;
+
+    av_freep(&s->decoded_buffer);
+
+    return 0;
+}
+
+static const AVOption options[] = {
+{ "use_buggy_lpc", "emulate old buggy lavc behavior", offsetof(FLACContext, buggy_lpc), AV_OPT_TYPE_BOOL, {.i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
+{ NULL },
+};
+
+static const AVClass flac_decoder_class = {
+    .class_name = "FLAC decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+const FFCodec ff_flac_decoder = {
+    .p.name         = "flac",
+    CODEC_LONG_NAME("FLAC (Free Lossless Audio Codec)"),
+    .p.type         = AVMEDIA_TYPE_AUDIO,
+    .p.id           = AV_CODEC_ID_FLAC,
+    .priv_data_size = sizeof(FLACContext),
+    .init           = flac_decode_init,
+    .close          = flac_decode_close,
+    FF_CODEC_DECODE_CB(flac_decode_frame),
+    .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
+                      AV_CODEC_CAP_DR1 |
+                      AV_CODEC_CAP_FRAME_THREADS,
+    .p.sample_fmts  = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+                                                      AV_SAMPLE_FMT_S16P,
+                                                      AV_SAMPLE_FMT_S32,
+                                                      AV_SAMPLE_FMT_S32P,
+                                                      AV_SAMPLE_FMT_NONE },
+    .p.priv_class   = &flac_decoder_class,
+};