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ffmpeg/libavcodec/alac.c

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/*
* 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 block_size;
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 * sizeof(*output_buffer));
output_count += 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 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) * sizeof(*buffer_out));
return;
}
if (predictor_coef_num == 31) {
/* simple 1st-order prediction */
if (output_size <= 1)
return;
for (i = 1; i < output_size; i++) {
buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],
readsamplesize);
}
return;
}
/* read warm-up samples */
for (i = 0; i < predictor_coef_num; i++) {
buffer_out[i + 1] = sign_extend(buffer_out[i] + error_buffer[i + 1],
readsamplesize);
}
/* NOTE: 4 and 8 are very common cases that could be optimized. */
/* general case */
for (i = predictor_coef_num + 1; i < output_size; i++) {
int j;
int val = 0;
int error_val = error_buffer[i];
int error_sign;
for (j = 0; j < predictor_coef_num; j++) {
val += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
predictor_coef_table[j];
}
val = (val + (1 << (predictor_quantitization - 1))) >>
predictor_quantitization;
val += buffer_out[0] + error_val;
buffer_out[predictor_coef_num + 1] = sign_extend(val, readsamplesize);
/* adapt LPC coefficients */
error_sign = sign_only(error_val);
if (error_sign) {
for (j = predictor_coef_num - 1; j >= 0 && error_val * error_sign > 0; j--) {
int sign;
val = buffer_out[0] - buffer_out[predictor_coef_num - j];
sign = sign_only(val) * error_sign;
predictor_coef_table[j] -= sign;
val *= sign;
error_val -= ((val >> predictor_quantitization) *
(predictor_coef_num - j));
}
}
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)"),
};
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