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-rw-r--r--media/ffvpx/libavcodec/flacdec.c672
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,
+};
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-25 10:07:42 +0000