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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 "libavutil/channel_layout.h"
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "unary.h"
#include "mathops.h"
#define ALAC_EXTRADATA_SIZE 36
#define MAX_CHANNELS 8
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
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;
enum RawDataBlockType {
/* At the moment, only SCE, CPE, LFE, and END are recognized. */
TYPE_SCE,
TYPE_CPE,
TYPE_CCE,
TYPE_LFE,
TYPE_DSE,
TYPE_PCE,
TYPE_FIL,
TYPE_END
};
static const uint8_t alac_channel_layout_offsets[8][8] = {
{ 0 },
{ 0, 1 },
{ 2, 0, 1 },
{ 2, 0, 1, 3 },
{ 2, 0, 1, 3, 4 },
{ 2, 0, 1, 4, 5, 3 },
{ 2, 0, 1, 4, 5, 6, 3 },
{ 2, 6, 7, 0, 1, 4, 5, 3 }
};
static const uint16_t alac_channel_layouts[8] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
AV_CH_LAYOUT_6POINT1_BACK,
AV_CH_LAYOUT_7POINT1_WIDE_BACK
};
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, void *data, 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: %d\n",
output_samples);
return AVERROR_INVALIDDATA;
}
if (!alac->nb_samples) {
/* get output buffer */
alac->frame.nb_samples = output_samples;
if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 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: %u != %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 *)alac->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];
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);
/* 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 *)alac->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;
enum RawDataBlockType 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) {
av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
return AVERROR_INVALIDDATA;
}
ret = decode_element(avctx, data,
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 (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;
*(AVFrame *)data = alac->frame;
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) {
av_log(alac->avctx, AV_LOG_ERROR, "max samples per frame invalid: %u\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: 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_S16P;
break;
case 24:
case 32: avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
break;
default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
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 > 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;
}
avctx->channel_layout = alac_channel_layouts[alac->channels - 1];
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 = AV_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)"),
};