xenia-project/FFmpeg
1
0
Fork 0
mirror of https://github.com/xenia-project/FFmpeg.git synced 2024-11-24 12:09:55 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity
eadd66a4af
FFmpeg/libavcodec/alac.c
Christophe Gisquet 30cac8313a alac: add option to decoded incorrect ALAC 
Prior to 56.1.100, incorrect ALAC files for 24bps content were produced, in
particular not decoding losslessly.

Add an option to allow correctly decoding those streams.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-18 21:59:23 +02:00

668 lines
22 KiB
C
Raw Blame History

/*
* 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 (10)
* 8bit rice param limit (14)
* 8bit channels
* 16bit maxRun (255)
* 32bit max coded frame size (0 means unknown)
* 32bit average bitrate (0 means unknown)
* 32bit samplerate
*/
#include <inttypes.h>
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "internal.h"
#include "thread.h"
#include "unary.h"
#include "mathops.h"
#include "alac_data.h"
#define ALAC_EXTRADATA_SIZE 36
typedef struct {
AVClass *class;
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 */
int direct_output;
int extra_bit_bug;
} 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 int 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;
if(get_bits_left(&alac->gb) <= 0)
return -1;
/* 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;
}
}
return 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 < nb_samples; 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 > 32U) {
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) {
ThreadFrame tframe = { .f = frame };
/* get output buffer */
frame->nb_samples = output_samples;
if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0)
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->direct_output) {
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];
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++) {
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=rice_decompress(alac, alac->predict_error_buffer[ch],
alac->nb_samples, bps,
rice_history_mult[ch] * alac->rice_history_mult / 4);
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.
*/
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++) {
if(get_bits_left(&alac->gb) <= 0)
return -1;
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 (alac->extra_bits && alac->extra_bit_bug) {
append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
alac->extra_bits, channels, alac->nb_samples);
}
if (channels == 2 && decorr_left_weight) {
decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,
decorr_shift, decorr_left_weight);
}
if (alac->extra_bits && !alac->extra_bit_bug) {
append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
alac->extra_bits, channels, alac->nb_samples);
}
if(av_sample_fmt_is_planar(avctx->sample_fmt)) {
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;
}
}else{
switch(alac->sample_size) {
case 16: {
int16_t *outbuffer = ((int16_t *)frame->extended_data[0]) + ch_index;
for (i = 0; i < alac->nb_samples; i++) {
for (ch = 0; ch < channels; ch++)
*outbuffer++ = alac->output_samples_buffer[ch][i];
outbuffer += alac->channels - channels;
}
}
break;
case 24: {
int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index;
for (i = 0; i < alac->nb_samples; i++) {
for (ch = 0; ch < channels; ch++)
*outbuffer++ = alac->output_samples_buffer[ch][i] << 8;
outbuffer += alac->channels - channels;
}
}
break;
case 32: {
int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index;
for (i = 0; i < alac->nb_samples; i++) {
for (ch = 0; ch < channels; ch++)
*outbuffer++ = alac->output_samples_buffer[ch][i];
outbuffer += alac->channels - channels;
}
}
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;
if ((ret = init_get_bits8(&alac->gb, avpkt->data, avpkt->size)) < 0)
return ret;
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\n", 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 (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));
}
if (alac->channels == ch)
*got_frame_ptr = 1;
else
av_log(avctx, AV_LOG_WARNING, "Failed to decode all channels\n");
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->direct_output)
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);
alac->direct_output = alac->sample_size > 16 && av_sample_fmt_is_planar(alac->avctx->sample_fmt);
if (!alac->direct_output) {
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;
int req_packed;
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, "extradata is too small\n");
return AVERROR_INVALIDDATA;
}
if (alac_set_info(alac)) {
av_log(avctx, AV_LOG_ERROR, "set_info failed\n");
return -1;
}
req_packed = LIBAVCODEC_VERSION_MAJOR < 55 && !av_sample_fmt_is_planar(avctx->request_sample_fmt);
switch (alac->sample_size) {
case 16: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_S16P;
break;
case 24:
case 32: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S32 : 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 || avctx->channels <= 0 ) {
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;
}
static int init_thread_copy(AVCodecContext *avctx)
{
ALACContext *alac = avctx->priv_data;
alac->avctx = avctx;
return allocate_buffers(alac);
}
static const AVOption options[] = {
{ "extra_bits_bug", "Force non-standard decoding process",
offsetof(ALACContext, extra_bit_bug), AV_OPT_TYPE_INT, { .i64 = 0 },
0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass alac_class = {
.class_name = "alac",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
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,
.init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy),
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS,
.priv_class = &alac_class
};
Reference in New Issue View Git Blame Copy Permalink