7e6593e977
x_modified is just unnecessary, and final_val can be removed by simplifying the unsigned-to-signed conversion.
635 lines
21 KiB
C
635 lines
21 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:
|
|
*
|
|
* 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 rice param limit (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 channels;
|
|
|
|
/* buffers */
|
|
int32_t *predict_error_buffer[MAX_CHANNELS];
|
|
int32_t *output_samples_buffer[MAX_CHANNELS];
|
|
int32_t *extra_bits_buffer[MAX_CHANNELS];
|
|
|
|
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 */
|
|
} ALACContext;
|
|
|
|
static inline int decode_scalar(GetBitContext *gb, int k, int readsamplesize)
|
|
{
|
|
int x = get_unary_0_9(gb);
|
|
|
|
if (x > 8) { /* RICE THRESHOLD */
|
|
/* use alternative encoding */
|
|
x = get_bits(gb, readsamplesize);
|
|
} 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 bastardized_rice_decompress(ALACContext *alac,
|
|
int32_t *output_buffer,
|
|
int output_size,
|
|
int readsamplesize,
|
|
int rice_history_mult)
|
|
{
|
|
int output_count;
|
|
unsigned int history = alac->rice_initial_history;
|
|
int sign_modifier = 0;
|
|
|
|
for (output_count = 0; output_count < output_size; output_count++) {
|
|
int x, k;
|
|
|
|
/* read k, that is bits as is */
|
|
k = av_log2((history >> 9) + 3);
|
|
k = FFMIN(k, alac->rice_limit);
|
|
x = decode_scalar(&alac->gb, k, readsamplesize);
|
|
x += sign_modifier;
|
|
sign_modifier = 0;
|
|
|
|
output_buffer[output_count] = (x >> 1) ^ -(x & 1);
|
|
|
|
/* now 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) && (output_count+1 < output_size)) {
|
|
int k;
|
|
unsigned int block_size;
|
|
|
|
sign_modifier = 1;
|
|
|
|
k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
|
|
k = FFMIN(k, alac->rice_limit);
|
|
|
|
block_size = decode_scalar(&alac->gb, k, 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;
|
|
}
|
|
}
|
|
}
|
|
|
|
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 == 31) {
|
|
/* simple 1st-order prediction */
|
|
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;
|
|
}
|
|
|
|
/* NOTE: 4 and 8 are very common cases that could be 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 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;
|
|
}
|
|
|
|
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 */
|
|
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->max_samples_per_frame) {
|
|
av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n",
|
|
outputsamples, alac->max_samples_per_frame);
|
|
return -1;
|
|
}
|
|
} else
|
|
outputsamples = alac->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->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++) {
|
|
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++) {
|
|
bastardized_rice_decompress(alac,
|
|
alac->predict_error_buffer[ch],
|
|
outputsamples,
|
|
readsamplesize,
|
|
ricemodifier[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.
|
|
*/
|
|
predictor_decompress_fir_adapt(alac->predict_error_buffer[ch],
|
|
alac->predict_error_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->predict_error_buffer[ch],
|
|
alac->output_samples_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++) {
|
|
for (ch = 0; ch < channels; ch++) {
|
|
alac->output_samples_buffer[ch][i] = get_sbits_long(&alac->gb,
|
|
alac->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->output_samples_buffer, outputsamples,
|
|
interlacing_shift, interlacing_leftweight);
|
|
}
|
|
|
|
if (alac->extra_bits) {
|
|
append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
|
|
alac->extra_bits, alac->channels, outputsamples);
|
|
}
|
|
|
|
switch(alac->sample_size) {
|
|
case 16:
|
|
if (channels == 2) {
|
|
interleave_stereo_16(alac->output_samples_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->output_samples_buffer[0][i];
|
|
}
|
|
}
|
|
break;
|
|
case 24:
|
|
if (channels == 2) {
|
|
interleave_stereo_24(alac->output_samples_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->output_samples_buffer[0][i] << 8;
|
|
}
|
|
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->channels; ch++) {
|
|
av_freep(&alac->predict_error_buffer[ch]);
|
|
av_freep(&alac->output_samples_buffer[ch]);
|
|
av_freep(&alac->extra_bits_buffer[ch]);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int allocate_buffers(ALACContext *alac)
|
|
{
|
|
int ch;
|
|
for (ch = 0; ch < alac->channels; ch++) {
|
|
int buf_size = alac->max_samples_per_frame * sizeof(int32_t);
|
|
|
|
FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],
|
|
buf_size, buf_alloc_fail);
|
|
|
|
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 >= UINT_MAX/4){
|
|
av_log(alac->avctx, AV_LOG_ERROR,
|
|
"max_samples_per_frame too large\n");
|
|
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: 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->sample_size) {
|
|
case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
|
|
break;
|
|
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->sample_size);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
|
|
if (alac->channels < 1) {
|
|
av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
|
|
alac->channels = avctx->channels;
|
|
} else {
|
|
if (alac->channels > MAX_CHANNELS)
|
|
alac->channels = avctx->channels;
|
|
else
|
|
avctx->channels = alac->channels;
|
|
}
|
|
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)"),
|
|
};
|