123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168169170171172173174175176177178179180181182183184185186187188189190191192193194195196197198199200201202203204205206207208209210211212213214215216217218219220221222223224225226227228229230231232233234235236237238239240241242243244245246247248249250251252253254255256257258259260261262263264265266267268269270271272273274275276277278279280281282283284285286287288289290291292293294295296297298299300301302303304305306307308309310311312313314315316317318319320321322323324325326327328329330331332333334335336337338339340341342343344345346347348349350351352353354355356357358359360361362363364365366367368369370371372373374375376377378379380381382383384385386387388389390391392393394395396397398399400401402403404405406407408409410411412413414415416417418419420421422423424425426427428429430431432433434435436437438439440441442443444445446447448449450451452453454455456457458459460461462463464465466467468469470471472473474475476477478479480481482483484485486487488489490491492493494495496497498499500501502503504505506507508509510511512513514515516517518519520521522523524525526527528529530531532533534535536537538539540541542543544545546547548549550551552553554555556557558559560561562563564565566567568569570571572573574575576577578579580581582583584585586587588589590591592593594595596597598599600601602603604605606607608609610611612613614615616617618619620621622623624625626627628629630631632633634635636637638639640641642643644645646647648649650651652653654655656657658659660661662663664665666667668669670671672673674675676677678679680681682683684685686687688689690691692693694695696697 |
- /*
- * ALAC (Apple Lossless Audio Codec) decoder
- * Copyright (c) 2005 David Hammerton
- *
- * 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
- * ALAC (Apple Lossless Audio Codec) decoder
- * @author 2005 David Hammerton
- * @see http://crazney.net/programs/itunes/alac.html
- *
- * Note: This decoder expects a 36-byte QuickTime atom to be
- * passed through the extradata[_size] fields. This atom is tacked onto
- * the end of an 'alac' stsd atom and has the following format:
- *
- * 32bit atom size
- * 32bit tag ("alac")
- * 32bit tag version (0)
- * 32bit samples per frame (used when not set explicitly in the frames)
- * 8bit compatible version (0)
- * 8bit sample size
- * 8bit history mult (40)
- * 8bit initial history (14)
- * 8bit kmodifier (10)
- * 8bit channels
- * 16bit maxRun (255)
- * 32bit max coded frame size (0 means unknown)
- * 32bit average bitrate (0 means unknown)
- * 32bit samplerate
- */
- #include "avcodec.h"
- #include "get_bits.h"
- #include "bytestream.h"
- #include "unary.h"
- #include "mathops.h"
- #define ALAC_EXTRADATA_SIZE 36
- #define MAX_CHANNELS 2
- typedef struct {
- AVCodecContext *avctx;
- AVFrame frame;
- GetBitContext gb;
- int numchannels;
- /* buffers */
- int32_t *predicterror_buffer[MAX_CHANNELS];
- int32_t *outputsamples_buffer[MAX_CHANNELS];
- int32_t *extra_bits_buffer[MAX_CHANNELS];
- /* stuff from setinfo */
- uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
- uint8_t setinfo_sample_size; /* 0x10 */
- uint8_t setinfo_rice_historymult; /* 0x28 */
- uint8_t setinfo_rice_initialhistory; /* 0x0a */
- uint8_t setinfo_rice_kmodifier; /* 0x0e */
- /* end setinfo stuff */
- int extra_bits; /**< number of extra bits beyond 16-bit */
- } ALACContext;
- static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
- /* read x - number of 1s before 0 represent the rice */
- int x = get_unary_0_9(gb);
- if (x > 8) { /* RICE THRESHOLD */
- /* use alternative encoding */
- x = get_bits(gb, readsamplesize);
- } else {
- if (k >= limit)
- k = limit;
- if (k != 1) {
- int extrabits = show_bits(gb, k);
- /* multiply x by 2^k - 1, as part of their strange algorithm */
- x = (x << k) - x;
- if (extrabits > 1) {
- x += extrabits - 1;
- skip_bits(gb, k);
- } else
- skip_bits(gb, k - 1);
- }
- }
- return x;
- }
- static int bastardized_rice_decompress(ALACContext *alac,
- int32_t *output_buffer,
- int output_size,
- int readsamplesize, /* arg_10 */
- int rice_initialhistory, /* arg424->b */
- int rice_kmodifier, /* arg424->d */
- int rice_historymult, /* arg424->c */
- int rice_kmodifier_mask /* arg424->e */
- )
- {
- int output_count;
- unsigned int history = rice_initialhistory;
- int sign_modifier = 0;
- for (output_count = 0; output_count < output_size; output_count++) {
- int32_t x;
- int32_t x_modified;
- int32_t final_val;
- /* standard rice encoding */
- int k; /* size of extra bits */
- if(get_bits_left(&alac->gb) <= 0)
- return -1;
- /* read k, that is bits as is */
- k = av_log2((history >> 9) + 3);
- x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
- x_modified = sign_modifier + x;
- final_val = (x_modified + 1) / 2;
- if (x_modified & 1) final_val *= -1;
- output_buffer[output_count] = final_val;
- sign_modifier = 0;
- /* now update the history */
- history += x_modified * rice_historymult
- - ((history * rice_historymult) >> 9);
- if (x_modified > 0xffff)
- history = 0xffff;
- /* special case: there may be compressed blocks of 0 */
- if ((history < 128) && (output_count+1 < output_size)) {
- int k;
- unsigned int block_size;
- sign_modifier = 1;
- k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
- block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
- if (block_size > 0) {
- if(block_size >= output_size - output_count){
- av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
- block_size= output_size - output_count - 1;
- }
- memset(&output_buffer[output_count+1], 0, block_size * 4);
- output_count += block_size;
- }
- if (block_size > 0xffff)
- sign_modifier = 0;
- history = 0;
- }
- }
- return 0;
- }
- static inline int sign_only(int v)
- {
- return v ? FFSIGN(v) : 0;
- }
- static void predictor_decompress_fir_adapt(int32_t *error_buffer,
- int32_t *buffer_out,
- int output_size,
- int readsamplesize,
- int16_t *predictor_coef_table,
- int predictor_coef_num,
- int predictor_quantitization)
- {
- int i;
- /* first sample always copies */
- *buffer_out = *error_buffer;
- if (!predictor_coef_num) {
- if (output_size <= 1)
- return;
- memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
- return;
- }
- if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
- /* second-best case scenario for fir decompression,
- * error describes a small difference from the previous sample only
- */
- if (output_size <= 1)
- return;
- for (i = 0; i < output_size - 1; i++) {
- int32_t prev_value;
- int32_t error_value;
- prev_value = buffer_out[i];
- error_value = error_buffer[i+1];
- buffer_out[i+1] =
- sign_extend((prev_value + error_value), readsamplesize);
- }
- return;
- }
- /* read warm-up samples */
- if (predictor_coef_num > 0)
- for (i = 0; i < predictor_coef_num; i++) {
- int32_t val;
- val = buffer_out[i] + error_buffer[i+1];
- val = sign_extend(val, readsamplesize);
- buffer_out[i+1] = val;
- }
- /* 4 and 8 are very common cases (the only ones i've seen). these
- * should be unrolled and optimized
- */
- /* general case */
- if (predictor_coef_num > 0) {
- for (i = predictor_coef_num + 1; i < output_size; i++) {
- int j;
- int sum = 0;
- int outval;
- int error_val = error_buffer[i];
- for (j = 0; j < predictor_coef_num; j++) {
- sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
- predictor_coef_table[j];
- }
- outval = (1 << (predictor_quantitization-1)) + sum;
- outval = outval >> predictor_quantitization;
- outval = outval + buffer_out[0] + error_val;
- outval = sign_extend(outval, readsamplesize);
- buffer_out[predictor_coef_num+1] = outval;
- if (error_val > 0) {
- int predictor_num = predictor_coef_num - 1;
- while (predictor_num >= 0 && error_val > 0) {
- int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
- int sign = sign_only(val);
- predictor_coef_table[predictor_num] -= sign;
- val *= sign; /* absolute value */
- error_val -= ((val >> predictor_quantitization) *
- (predictor_coef_num - predictor_num));
- predictor_num--;
- }
- } else if (error_val < 0) {
- int predictor_num = predictor_coef_num - 1;
- while (predictor_num >= 0 && error_val < 0) {
- int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
- int sign = - sign_only(val);
- predictor_coef_table[predictor_num] -= sign;
- val *= sign; /* neg value */
- error_val -= ((val >> predictor_quantitization) *
- (predictor_coef_num - predictor_num));
- predictor_num--;
- }
- }
- buffer_out++;
- }
- }
- }
- static void decorrelate_stereo(int32_t *buffer[MAX_CHANNELS],
- int numsamples, uint8_t interlacing_shift,
- uint8_t interlacing_leftweight)
- {
- int i;
- for (i = 0; i < numsamples; i++) {
- int32_t a, b;
- a = buffer[0][i];
- b = buffer[1][i];
- a -= (b * interlacing_leftweight) >> interlacing_shift;
- b += a;
- buffer[0][i] = b;
- buffer[1][i] = a;
- }
- }
- static void append_extra_bits(int32_t *buffer[MAX_CHANNELS],
- int32_t *extra_bits_buffer[MAX_CHANNELS],
- int extra_bits, int numchannels, int numsamples)
- {
- int i, ch;
- for (ch = 0; ch < numchannels; ch++)
- for (i = 0; i < numsamples; i++)
- buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
- }
- static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS],
- int16_t *buffer_out, int numsamples)
- {
- int i;
- for (i = 0; i < numsamples; i++) {
- *buffer_out++ = buffer[0][i];
- *buffer_out++ = buffer[1][i];
- }
- }
- static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
- int32_t *buffer_out, int numsamples)
- {
- int i;
- for (i = 0; i < numsamples; i++) {
- *buffer_out++ = buffer[0][i] << 8;
- *buffer_out++ = buffer[1][i] << 8;
- }
- }
- static void interleave_stereo_32(int32_t *buffer[MAX_CHANNELS],
- int32_t *buffer_out, int numsamples)
- {
- int i;
- for (i = 0; i < numsamples; i++) {
- *buffer_out++ = buffer[0][i];
- *buffer_out++ = buffer[1][i];
- }
- }
- static int alac_decode_frame(AVCodecContext *avctx, void *data,
- int *got_frame_ptr, AVPacket *avpkt)
- {
- const uint8_t *inbuffer = avpkt->data;
- int input_buffer_size = avpkt->size;
- ALACContext *alac = avctx->priv_data;
- int channels;
- unsigned int outputsamples;
- int hassize;
- unsigned int readsamplesize;
- int isnotcompressed;
- uint8_t interlacing_shift;
- uint8_t interlacing_leftweight;
- int i, ch, ret;
- init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
- channels = get_bits(&alac->gb, 3) + 1;
- if (channels != avctx->channels) {
- av_log(avctx, AV_LOG_ERROR, "frame header channel count mismatch\n");
- return AVERROR_INVALIDDATA;
- }
- /* 2^result = something to do with output waiting.
- * perhaps matters if we read > 1 frame in a pass?
- */
- skip_bits(&alac->gb, 4);
- skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */
- /* the output sample size is stored soon */
- hassize = get_bits1(&alac->gb);
- alac->extra_bits = get_bits(&alac->gb, 2) << 3;
- /* whether the frame is compressed */
- isnotcompressed = get_bits1(&alac->gb);
- if (hassize) {
- /* now read the number of samples as a 32bit integer */
- outputsamples = get_bits_long(&alac->gb, 32);
- if(outputsamples > alac->setinfo_max_samples_per_frame){
- av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
- return -1;
- }
- } else
- outputsamples = alac->setinfo_max_samples_per_frame;
- /* get output buffer */
- if (outputsamples > INT32_MAX) {
- av_log(avctx, AV_LOG_ERROR, "unsupported block size: %u\n", outputsamples);
- return AVERROR_INVALIDDATA;
- }
- alac->frame.nb_samples = outputsamples;
- if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
- return ret;
- }
- readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1;
- if (readsamplesize > MIN_CACHE_BITS) {
- av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
- return -1;
- }
- if (!isnotcompressed) {
- /* so it is compressed */
- int16_t predictor_coef_table[MAX_CHANNELS][32];
- int predictor_coef_num[MAX_CHANNELS];
- int prediction_type[MAX_CHANNELS];
- int prediction_quantitization[MAX_CHANNELS];
- int ricemodifier[MAX_CHANNELS];
- interlacing_shift = get_bits(&alac->gb, 8);
- interlacing_leftweight = get_bits(&alac->gb, 8);
- for (ch = 0; ch < channels; ch++) {
- prediction_type[ch] = get_bits(&alac->gb, 4);
- prediction_quantitization[ch] = get_bits(&alac->gb, 4);
- ricemodifier[ch] = get_bits(&alac->gb, 3);
- predictor_coef_num[ch] = get_bits(&alac->gb, 5);
- /* read the predictor table */
- for (i = 0; i < predictor_coef_num[ch]; i++)
- predictor_coef_table[ch][i] = (int16_t)get_bits(&alac->gb, 16);
- }
- if (alac->extra_bits) {
- for (i = 0; i < outputsamples; i++) {
- if(get_bits_left(&alac->gb) <= 0)
- return -1;
- for (ch = 0; ch < channels; ch++)
- alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
- }
- }
- for (ch = 0; ch < channels; ch++) {
- int ret = bastardized_rice_decompress(alac,
- alac->predicterror_buffer[ch],
- outputsamples,
- readsamplesize,
- alac->setinfo_rice_initialhistory,
- alac->setinfo_rice_kmodifier,
- ricemodifier[ch] * alac->setinfo_rice_historymult / 4,
- (1 << alac->setinfo_rice_kmodifier) - 1);
- if(ret<0)
- return ret;
- /* adaptive FIR filter */
- if (prediction_type[ch] == 15) {
- /* Prediction type 15 runs the adaptive FIR twice.
- * The first pass uses the special-case coef_num = 31, while
- * the second pass uses the coefs from the bitstream.
- *
- * However, this prediction type is not currently used by the
- * reference encoder.
- */
- predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
- alac->predicterror_buffer[ch],
- outputsamples, readsamplesize,
- NULL, 31, 0);
- } else if (prediction_type[ch] > 0) {
- av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
- prediction_type[ch]);
- }
- predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
- alac->outputsamples_buffer[ch],
- outputsamples, readsamplesize,
- predictor_coef_table[ch],
- predictor_coef_num[ch],
- prediction_quantitization[ch]);
- }
- } else {
- /* not compressed, easy case */
- for (i = 0; i < outputsamples; i++) {
- if(get_bits_left(&alac->gb) <= 0)
- return -1;
- for (ch = 0; ch < channels; ch++) {
- alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb,
- alac->setinfo_sample_size);
- }
- }
- alac->extra_bits = 0;
- interlacing_shift = 0;
- interlacing_leftweight = 0;
- }
- if (get_bits(&alac->gb, 3) != 7)
- av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
- if (channels == 2 && interlacing_leftweight) {
- decorrelate_stereo(alac->outputsamples_buffer, outputsamples,
- interlacing_shift, interlacing_leftweight);
- }
- if (alac->extra_bits) {
- append_extra_bits(alac->outputsamples_buffer, alac->extra_bits_buffer,
- alac->extra_bits, alac->numchannels, outputsamples);
- }
- switch(alac->setinfo_sample_size) {
- case 16:
- if (channels == 2) {
- interleave_stereo_16(alac->outputsamples_buffer,
- (int16_t *)alac->frame.data[0], outputsamples);
- } else {
- int16_t *outbuffer = (int16_t *)alac->frame.data[0];
- for (i = 0; i < outputsamples; i++) {
- outbuffer[i] = alac->outputsamples_buffer[0][i];
- }
- }
- break;
- case 24:
- if (channels == 2) {
- interleave_stereo_24(alac->outputsamples_buffer,
- (int32_t *)alac->frame.data[0], outputsamples);
- } else {
- int32_t *outbuffer = (int32_t *)alac->frame.data[0];
- for (i = 0; i < outputsamples; i++)
- outbuffer[i] = alac->outputsamples_buffer[0][i] << 8;
- }
- break;
- case 32:
- if (channels == 2) {
- interleave_stereo_32(alac->outputsamples_buffer,
- (int32_t *)alac->frame.data[0], outputsamples);
- } else {
- int32_t *outbuffer = (int32_t *)alac->frame.data[0];
- for (i = 0; i < outputsamples; i++)
- outbuffer[i] = alac->outputsamples_buffer[0][i];
- }
- break;
- }
- if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
- av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
- *got_frame_ptr = 1;
- *(AVFrame *)data = alac->frame;
- return input_buffer_size;
- }
- static av_cold int alac_decode_close(AVCodecContext *avctx)
- {
- ALACContext *alac = avctx->priv_data;
- int ch;
- for (ch = 0; ch < alac->numchannels; ch++) {
- av_freep(&alac->predicterror_buffer[ch]);
- av_freep(&alac->outputsamples_buffer[ch]);
- av_freep(&alac->extra_bits_buffer[ch]);
- }
- return 0;
- }
- static int allocate_buffers(ALACContext *alac)
- {
- int ch;
- for (ch = 0; ch < alac->numchannels; ch++) {
- int buf_size = alac->setinfo_max_samples_per_frame * sizeof(int32_t);
- FF_ALLOC_OR_GOTO(alac->avctx, alac->predicterror_buffer[ch],
- buf_size, buf_alloc_fail);
- FF_ALLOC_OR_GOTO(alac->avctx, alac->outputsamples_buffer[ch],
- buf_size, buf_alloc_fail);
- FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
- buf_size, buf_alloc_fail);
- }
- return 0;
- buf_alloc_fail:
- alac_decode_close(alac->avctx);
- return AVERROR(ENOMEM);
- }
- static int alac_set_info(ALACContext *alac)
- {
- GetByteContext gb;
- bytestream2_init(&gb, alac->avctx->extradata,
- alac->avctx->extradata_size);
- bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
- /* buffer size / 2 ? */
- alac->setinfo_max_samples_per_frame = bytestream2_get_be32u(&gb);
- if (alac->setinfo_max_samples_per_frame >= UINT_MAX/4){
- av_log(alac->avctx, AV_LOG_ERROR,
- "setinfo_max_samples_per_frame too large\n");
- return AVERROR_INVALIDDATA;
- }
- bytestream2_skipu(&gb, 1); // compatible version
- alac->setinfo_sample_size = bytestream2_get_byteu(&gb);
- alac->setinfo_rice_historymult = bytestream2_get_byteu(&gb);
- alac->setinfo_rice_initialhistory = bytestream2_get_byteu(&gb);
- alac->setinfo_rice_kmodifier = bytestream2_get_byteu(&gb);
- alac->numchannels = bytestream2_get_byteu(&gb);
- bytestream2_get_be16u(&gb); // maxRun
- bytestream2_get_be32u(&gb); // max coded frame size
- bytestream2_get_be32u(&gb); // average bitrate
- bytestream2_get_be32u(&gb); // samplerate
- return 0;
- }
- static av_cold int alac_decode_init(AVCodecContext * avctx)
- {
- int ret;
- ALACContext *alac = avctx->priv_data;
- alac->avctx = avctx;
- /* initialize from the extradata */
- if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
- av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
- ALAC_EXTRADATA_SIZE);
- return -1;
- }
- if (alac_set_info(alac)) {
- av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
- return -1;
- }
- switch (alac->setinfo_sample_size) {
- case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- break;
- case 32:
- case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
- break;
- default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
- alac->setinfo_sample_size);
- return AVERROR_PATCHWELCOME;
- }
- if (alac->numchannels < 1) {
- av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
- alac->numchannels = avctx->channels;
- } else {
- if (alac->numchannels > MAX_CHANNELS)
- alac->numchannels = avctx->channels;
- else
- avctx->channels = alac->numchannels;
- }
- if (avctx->channels > MAX_CHANNELS) {
- av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
- avctx->channels);
- return AVERROR_PATCHWELCOME;
- }
- if ((ret = allocate_buffers(alac)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
- return ret;
- }
- avcodec_get_frame_defaults(&alac->frame);
- avctx->coded_frame = &alac->frame;
- return 0;
- }
- AVCodec ff_alac_decoder = {
- .name = "alac",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_ALAC,
- .priv_data_size = sizeof(ALACContext),
- .init = alac_decode_init,
- .close = alac_decode_close,
- .decode = alac_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
- };
|