mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
796dca027b
If we encounter an END element before anything is decoded, we would return success even though the output frame has not been allocated, which is invalid. CC: libav-stable@libav.org Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
596 lines
19 KiB
C
596 lines
19 KiB
C
/*
|
|
* ALAC (Apple Lossless Audio Codec) decoder
|
|
* Copyright (c) 2005 David Hammerton
|
|
*
|
|
* This file is part of Libav.
|
|
*
|
|
* Libav 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.
|
|
*
|
|
* Libav 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 Libav; 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:
|
|
*
|
|
* 32 bits atom size
|
|
* 32 bits tag ("alac")
|
|
* 32 bits tag version (0)
|
|
* 32 bits samples per frame (used when not set explicitly in the frames)
|
|
* 8 bits compatible version (0)
|
|
* 8 bits sample size
|
|
* 8 bits history mult (40)
|
|
* 8 bits initial history (14)
|
|
* 8 bits rice param limit (10)
|
|
* 8 bits channels
|
|
* 16 bits maxRun (255)
|
|
* 32 bits max coded frame size (0 means unknown)
|
|
* 32 bits average bitrate (0 means unknown)
|
|
* 32 bits samplerate
|
|
*/
|
|
|
|
#include <inttypes.h>
|
|
|
|
#include "libavutil/channel_layout.h"
|
|
#include "avcodec.h"
|
|
#include "get_bits.h"
|
|
#include "bytestream.h"
|
|
#include "internal.h"
|
|
#include "unary.h"
|
|
#include "mathops.h"
|
|
#include "alac_data.h"
|
|
|
|
#define ALAC_EXTRADATA_SIZE 36
|
|
|
|
typedef struct ALACContext {
|
|
AVCodecContext *avctx;
|
|
GetBitContext gb;
|
|
int channels;
|
|
|
|
int32_t *predict_error_buffer[2];
|
|
int32_t *output_samples_buffer[2];
|
|
int32_t *extra_bits_buffer[2];
|
|
|
|
uint32_t max_samples_per_frame;
|
|
uint8_t sample_size;
|
|
uint8_t rice_history_mult;
|
|
uint8_t rice_initial_history;
|
|
uint8_t rice_limit;
|
|
|
|
int extra_bits; /**< number of extra bits beyond 16-bit */
|
|
int nb_samples; /**< number of samples in the current frame */
|
|
} ALACContext;
|
|
|
|
static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
|
|
{
|
|
unsigned int x = get_unary_0_9(gb);
|
|
|
|
if (x > 8) { /* RICE THRESHOLD */
|
|
/* use alternative encoding */
|
|
x = get_bits_long(gb, bps);
|
|
} else 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 void rice_decompress(ALACContext *alac, int32_t *output_buffer,
|
|
int nb_samples, int bps, int rice_history_mult)
|
|
{
|
|
int i;
|
|
unsigned int history = alac->rice_initial_history;
|
|
int sign_modifier = 0;
|
|
|
|
for (i = 0; i < nb_samples; i++) {
|
|
int k;
|
|
unsigned int x;
|
|
|
|
/* calculate rice param and decode next value */
|
|
k = av_log2((history >> 9) + 3);
|
|
k = FFMIN(k, alac->rice_limit);
|
|
x = decode_scalar(&alac->gb, k, bps);
|
|
x += sign_modifier;
|
|
sign_modifier = 0;
|
|
output_buffer[i] = (x >> 1) ^ -(x & 1);
|
|
|
|
/* update the history */
|
|
if (x > 0xffff)
|
|
history = 0xffff;
|
|
else
|
|
history += x * rice_history_mult -
|
|
((history * rice_history_mult) >> 9);
|
|
|
|
/* special case: there may be compressed blocks of 0 */
|
|
if ((history < 128) && (i + 1 < nb_samples)) {
|
|
int block_size;
|
|
|
|
/* calculate rice param and decode block size */
|
|
k = 7 - av_log2(history) + ((history + 16) >> 6);
|
|
k = FFMIN(k, alac->rice_limit);
|
|
block_size = decode_scalar(&alac->gb, k, 16);
|
|
|
|
if (block_size > 0) {
|
|
if (block_size >= nb_samples - i) {
|
|
av_log(alac->avctx, AV_LOG_ERROR,
|
|
"invalid zero block size of %d %d %d\n", block_size,
|
|
nb_samples, i);
|
|
block_size = nb_samples - i - 1;
|
|
}
|
|
memset(&output_buffer[i + 1], 0,
|
|
block_size * sizeof(*output_buffer));
|
|
i += block_size;
|
|
}
|
|
if (block_size <= 0xffff)
|
|
sign_modifier = 1;
|
|
history = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
static inline int sign_only(int v)
|
|
{
|
|
return v ? FFSIGN(v) : 0;
|
|
}
|
|
|
|
static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
|
|
int nb_samples, int bps, int16_t *lpc_coefs,
|
|
int lpc_order, int lpc_quant)
|
|
{
|
|
int i;
|
|
int32_t *pred = buffer_out;
|
|
|
|
/* first sample always copies */
|
|
*buffer_out = *error_buffer;
|
|
|
|
if (nb_samples <= 1)
|
|
return;
|
|
|
|
if (!lpc_order) {
|
|
memcpy(&buffer_out[1], &error_buffer[1],
|
|
(nb_samples - 1) * sizeof(*buffer_out));
|
|
return;
|
|
}
|
|
|
|
if (lpc_order == 31) {
|
|
/* simple 1st-order prediction */
|
|
for (i = 1; i < nb_samples; i++) {
|
|
buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],
|
|
bps);
|
|
}
|
|
return;
|
|
}
|
|
|
|
/* read warm-up samples */
|
|
for (i = 1; i <= lpc_order; i++)
|
|
buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps);
|
|
|
|
/* NOTE: 4 and 8 are very common cases that could be optimized. */
|
|
|
|
for (; i < nb_samples; i++) {
|
|
int j;
|
|
int val = 0;
|
|
int error_val = error_buffer[i];
|
|
int error_sign;
|
|
int d = *pred++;
|
|
|
|
/* LPC prediction */
|
|
for (j = 0; j < lpc_order; j++)
|
|
val += (pred[j] - d) * lpc_coefs[j];
|
|
val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
|
|
val += d + error_val;
|
|
buffer_out[i] = sign_extend(val, bps);
|
|
|
|
/* adapt LPC coefficients */
|
|
error_sign = sign_only(error_val);
|
|
if (error_sign) {
|
|
for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
|
|
int sign;
|
|
val = d - pred[j];
|
|
sign = sign_only(val) * error_sign;
|
|
lpc_coefs[j] -= sign;
|
|
val *= sign;
|
|
error_val -= (val >> lpc_quant) * (j + 1);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void decorrelate_stereo(int32_t *buffer[2], int nb_samples,
|
|
int decorr_shift, int decorr_left_weight)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < nb_samples; i++) {
|
|
int32_t a, b;
|
|
|
|
a = buffer[0][i];
|
|
b = buffer[1][i];
|
|
|
|
a -= (b * decorr_left_weight) >> decorr_shift;
|
|
b += a;
|
|
|
|
buffer[0][i] = b;
|
|
buffer[1][i] = a;
|
|
}
|
|
}
|
|
|
|
static void append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2],
|
|
int extra_bits, int channels, int nb_samples)
|
|
{
|
|
int i, ch;
|
|
|
|
for (ch = 0; ch < channels; ch++)
|
|
for (i = 0; i < nb_samples; i++)
|
|
buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
|
|
}
|
|
|
|
static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
|
|
int channels)
|
|
{
|
|
ALACContext *alac = avctx->priv_data;
|
|
int has_size, bps, is_compressed, decorr_shift, decorr_left_weight, ret;
|
|
uint32_t output_samples;
|
|
int i, ch;
|
|
|
|
skip_bits(&alac->gb, 4); /* element instance tag */
|
|
skip_bits(&alac->gb, 12); /* unused header bits */
|
|
|
|
/* the number of output samples is stored in the frame */
|
|
has_size = get_bits1(&alac->gb);
|
|
|
|
alac->extra_bits = get_bits(&alac->gb, 2) << 3;
|
|
bps = alac->sample_size - alac->extra_bits + channels - 1;
|
|
if (bps > 32) {
|
|
av_log(avctx, AV_LOG_ERROR, "bps is unsupported: %d\n", bps);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
|
|
/* whether the frame is compressed */
|
|
is_compressed = !get_bits1(&alac->gb);
|
|
|
|
if (has_size)
|
|
output_samples = get_bits_long(&alac->gb, 32);
|
|
else
|
|
output_samples = alac->max_samples_per_frame;
|
|
if (!output_samples || output_samples > alac->max_samples_per_frame) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %"PRIu32"\n",
|
|
output_samples);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (!alac->nb_samples) {
|
|
/* get output buffer */
|
|
frame->nb_samples = output_samples;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
|
return ret;
|
|
}
|
|
} else if (output_samples != alac->nb_samples) {
|
|
av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %"PRIu32" != %d\n",
|
|
output_samples, alac->nb_samples);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
alac->nb_samples = output_samples;
|
|
if (alac->sample_size > 16) {
|
|
for (ch = 0; ch < channels; ch++)
|
|
alac->output_samples_buffer[ch] = (int32_t *)frame->extended_data[ch_index + ch];
|
|
}
|
|
|
|
if (is_compressed) {
|
|
int16_t lpc_coefs[2][32];
|
|
int lpc_order[2];
|
|
int prediction_type[2];
|
|
int lpc_quant[2];
|
|
int rice_history_mult[2];
|
|
|
|
if (!alac->rice_limit) {
|
|
avpriv_request_sample(alac->avctx,
|
|
"Compression with rice limit 0");
|
|
return AVERROR(ENOSYS);
|
|
}
|
|
|
|
decorr_shift = get_bits(&alac->gb, 8);
|
|
decorr_left_weight = get_bits(&alac->gb, 8);
|
|
|
|
for (ch = 0; ch < channels; ch++) {
|
|
prediction_type[ch] = get_bits(&alac->gb, 4);
|
|
lpc_quant[ch] = get_bits(&alac->gb, 4);
|
|
rice_history_mult[ch] = get_bits(&alac->gb, 3);
|
|
lpc_order[ch] = get_bits(&alac->gb, 5);
|
|
|
|
if (lpc_order[ch] >= alac->max_samples_per_frame)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
/* read the predictor table */
|
|
for (i = lpc_order[ch] - 1; i >= 0; i--)
|
|
lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);
|
|
}
|
|
|
|
if (alac->extra_bits) {
|
|
for (i = 0; i < alac->nb_samples; i++) {
|
|
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++) {
|
|
rice_decompress(alac, alac->predict_error_buffer[ch],
|
|
alac->nb_samples, bps,
|
|
rice_history_mult[ch] * alac->rice_history_mult / 4);
|
|
|
|
/* 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.
|
|
*/
|
|
lpc_prediction(alac->predict_error_buffer[ch],
|
|
alac->predict_error_buffer[ch],
|
|
alac->nb_samples, bps, NULL, 31, 0);
|
|
} else if (prediction_type[ch] > 0) {
|
|
av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
|
|
prediction_type[ch]);
|
|
}
|
|
lpc_prediction(alac->predict_error_buffer[ch],
|
|
alac->output_samples_buffer[ch], alac->nb_samples,
|
|
bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
|
|
}
|
|
} else {
|
|
/* not compressed, easy case */
|
|
for (i = 0; i < alac->nb_samples; i++) {
|
|
for (ch = 0; ch < channels; ch++) {
|
|
alac->output_samples_buffer[ch][i] =
|
|
get_sbits_long(&alac->gb, alac->sample_size);
|
|
}
|
|
}
|
|
alac->extra_bits = 0;
|
|
decorr_shift = 0;
|
|
decorr_left_weight = 0;
|
|
}
|
|
|
|
if (channels == 2 && decorr_left_weight) {
|
|
decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,
|
|
decorr_shift, decorr_left_weight);
|
|
}
|
|
|
|
if (alac->extra_bits) {
|
|
append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
|
|
alac->extra_bits, channels, alac->nb_samples);
|
|
}
|
|
|
|
switch(alac->sample_size) {
|
|
case 16: {
|
|
for (ch = 0; ch < channels; ch++) {
|
|
int16_t *outbuffer = (int16_t *)frame->extended_data[ch_index + ch];
|
|
for (i = 0; i < alac->nb_samples; i++)
|
|
*outbuffer++ = alac->output_samples_buffer[ch][i];
|
|
}}
|
|
break;
|
|
case 24: {
|
|
for (ch = 0; ch < channels; ch++) {
|
|
for (i = 0; i < alac->nb_samples; i++)
|
|
alac->output_samples_buffer[ch][i] <<= 8;
|
|
}}
|
|
break;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int alac_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
ALACContext *alac = avctx->priv_data;
|
|
AVFrame *frame = data;
|
|
enum AlacRawDataBlockType element;
|
|
int channels;
|
|
int ch, ret, got_end;
|
|
|
|
init_get_bits(&alac->gb, avpkt->data, avpkt->size * 8);
|
|
|
|
got_end = 0;
|
|
alac->nb_samples = 0;
|
|
ch = 0;
|
|
while (get_bits_left(&alac->gb) >= 3) {
|
|
element = get_bits(&alac->gb, 3);
|
|
if (element == TYPE_END) {
|
|
got_end = 1;
|
|
break;
|
|
}
|
|
if (element > TYPE_CPE && element != TYPE_LFE) {
|
|
av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d", element);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
|
|
channels = (element == TYPE_CPE) ? 2 : 1;
|
|
if (ch + channels > alac->channels ||
|
|
ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
ret = decode_element(avctx, frame,
|
|
ff_alac_channel_layout_offsets[alac->channels - 1][ch],
|
|
channels);
|
|
if (ret < 0 && get_bits_left(&alac->gb))
|
|
return ret;
|
|
|
|
ch += channels;
|
|
}
|
|
if (!got_end) {
|
|
av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (!alac->nb_samples) {
|
|
av_log(avctx, AV_LOG_ERROR, "No decodable data in the packet\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) {
|
|
av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n",
|
|
avpkt->size * 8 - get_bits_count(&alac->gb));
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return avpkt->size;
|
|
}
|
|
|
|
static av_cold int alac_decode_close(AVCodecContext *avctx)
|
|
{
|
|
ALACContext *alac = avctx->priv_data;
|
|
|
|
int ch;
|
|
for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
|
|
av_freep(&alac->predict_error_buffer[ch]);
|
|
if (alac->sample_size == 16)
|
|
av_freep(&alac->output_samples_buffer[ch]);
|
|
av_freep(&alac->extra_bits_buffer[ch]);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int allocate_buffers(ALACContext *alac)
|
|
{
|
|
int ch;
|
|
int buf_size = alac->max_samples_per_frame * sizeof(int32_t);
|
|
|
|
for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
|
|
FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],
|
|
buf_size, buf_alloc_fail);
|
|
|
|
if (alac->sample_size == 16) {
|
|
FF_ALLOC_OR_GOTO(alac->avctx, alac->output_samples_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
|
|
|
|
alac->max_samples_per_frame = bytestream2_get_be32u(&gb);
|
|
if (!alac->max_samples_per_frame ||
|
|
alac->max_samples_per_frame > INT_MAX / sizeof(int32_t)) {
|
|
av_log(alac->avctx, AV_LOG_ERROR,
|
|
"max samples per frame invalid: %"PRIu32"\n",
|
|
alac->max_samples_per_frame);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
bytestream2_skipu(&gb, 1); // compatible version
|
|
alac->sample_size = bytestream2_get_byteu(&gb);
|
|
alac->rice_history_mult = bytestream2_get_byteu(&gb);
|
|
alac->rice_initial_history = bytestream2_get_byteu(&gb);
|
|
alac->rice_limit = bytestream2_get_byteu(&gb);
|
|
alac->channels = 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: extradata is too small\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (alac_set_info(alac)) {
|
|
av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
|
|
return -1;
|
|
}
|
|
|
|
switch (alac->sample_size) {
|
|
case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
|
|
break;
|
|
case 24:
|
|
case 32: avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
|
|
break;
|
|
default: avpriv_request_sample(avctx, "Sample depth %d", alac->sample_size);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
avctx->bits_per_raw_sample = alac->sample_size;
|
|
|
|
if (alac->channels < 1) {
|
|
av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
|
|
alac->channels = avctx->channels;
|
|
} else {
|
|
if (alac->channels > ALAC_MAX_CHANNELS)
|
|
alac->channels = avctx->channels;
|
|
else
|
|
avctx->channels = alac->channels;
|
|
}
|
|
if (avctx->channels > ALAC_MAX_CHANNELS) {
|
|
av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
|
|
avctx->channels);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
avctx->channel_layout = ff_alac_channel_layouts[alac->channels - 1];
|
|
|
|
if ((ret = allocate_buffers(alac)) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_alac_decoder = {
|
|
.name = "alac",
|
|
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_ALAC,
|
|
.priv_data_size = sizeof(ALACContext),
|
|
.init = alac_decode_init,
|
|
.close = alac_decode_close,
|
|
.decode = alac_decode_frame,
|
|
.capabilities = AV_CODEC_CAP_DR1,
|
|
};
|