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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
/*
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* ALAC (Apple Lossless Audio Codec) decoder
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* Copyright (c) 2005 David Hammerton
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* ALAC (Apple Lossless Audio Codec) decoder
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* @author 2005 David Hammerton
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* @see http://crazney.net/programs/itunes/alac.html
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*
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* Note: This decoder expects a 36-byte QuickTime atom to be
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* passed through the extradata[_size] fields. This atom is tacked onto
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* the end of an 'alac' stsd atom and has the following format:
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*
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* 32 bits atom size
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* 32 bits tag ("alac")
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* 32 bits tag version (0)
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* 32 bits samples per frame (used when not set explicitly in the frames)
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* 8 bits compatible version (0)
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* 8 bits sample size
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* 8 bits history mult (40)
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* 8 bits initial history (14)
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* 8 bits rice param limit (10)
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* 8 bits channels
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* 16 bits maxRun (255)
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* 32 bits max coded frame size (0 means unknown)
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* 32 bits average bitrate (0 means unknown)
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* 32 bits samplerate
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*/
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#include <inttypes.h>
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#include "libavutil/channel_layout.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "bytestream.h"
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#include "internal.h"
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#include "unary.h"
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#include "mathops.h"
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#include "alac_data.h"
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#define ALAC_EXTRADATA_SIZE 36
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typedef struct ALACContext {
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AVCodecContext *avctx;
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GetBitContext gb;
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int channels;
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int32_t *predict_error_buffer[2];
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int32_t *output_samples_buffer[2];
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int32_t *extra_bits_buffer[2];
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uint32_t max_samples_per_frame;
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uint8_t sample_size;
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uint8_t rice_history_mult;
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uint8_t rice_initial_history;
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uint8_t rice_limit;
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int extra_bits; /**< number of extra bits beyond 16-bit */
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int nb_samples; /**< number of samples in the current frame */
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} ALACContext;
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static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
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{
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unsigned int x = get_unary_0_9(gb);
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if (x > 8) { /* RICE THRESHOLD */
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/* use alternative encoding */
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x = get_bits_long(gb, bps);
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} else if (k != 1) {
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int extrabits = show_bits(gb, k);
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/* multiply x by 2^k - 1, as part of their strange algorithm */
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x = (x << k) - x;
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if (extrabits > 1) {
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x += extrabits - 1;
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skip_bits(gb, k);
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} else
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skip_bits(gb, k - 1);
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}
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return x;
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}
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static void rice_decompress(ALACContext *alac, int32_t *output_buffer,
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int nb_samples, int bps, int rice_history_mult)
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{
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int i;
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unsigned int history = alac->rice_initial_history;
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int sign_modifier = 0;
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for (i = 0; i < nb_samples; i++) {
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int k;
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unsigned int x;
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/* calculate rice param and decode next value */
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k = av_log2((history >> 9) + 3);
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k = FFMIN(k, alac->rice_limit);
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x = decode_scalar(&alac->gb, k, bps);
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x += sign_modifier;
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sign_modifier = 0;
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output_buffer[i] = (x >> 1) ^ -(x & 1);
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/* update the history */
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if (x > 0xffff)
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history = 0xffff;
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else
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history += x * rice_history_mult -
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((history * rice_history_mult) >> 9);
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/* special case: there may be compressed blocks of 0 */
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if ((history < 128) && (i + 1 < nb_samples)) {
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int block_size;
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/* calculate rice param and decode block size */
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k = 7 - av_log2(history) + ((history + 16) >> 6);
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k = FFMIN(k, alac->rice_limit);
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block_size = decode_scalar(&alac->gb, k, 16);
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if (block_size > 0) {
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if (block_size >= nb_samples - i) {
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av_log(alac->avctx, AV_LOG_ERROR,
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"invalid zero block size of %d %d %d\n", block_size,
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nb_samples, i);
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block_size = nb_samples - i - 1;
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}
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memset(&output_buffer[i + 1], 0,
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block_size * sizeof(*output_buffer));
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i += block_size;
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}
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if (block_size <= 0xffff)
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sign_modifier = 1;
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history = 0;
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}
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}
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}
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static inline int sign_only(int v)
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{
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return v ? FFSIGN(v) : 0;
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}
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static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
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int nb_samples, int bps, int16_t *lpc_coefs,
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int lpc_order, int lpc_quant)
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{
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int i;
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int32_t *pred = buffer_out;
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/* first sample always copies */
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*buffer_out = *error_buffer;
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if (nb_samples <= 1)
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return;
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if (!lpc_order) {
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memcpy(&buffer_out[1], &error_buffer[1],
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(nb_samples - 1) * sizeof(*buffer_out));
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return;
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}
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if (lpc_order == 31) {
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/* simple 1st-order prediction */
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for (i = 1; i < nb_samples; i++) {
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buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],
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bps);
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}
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return;
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}
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/* read warm-up samples */
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for (i = 1; i <= lpc_order; i++)
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buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps);
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/* NOTE: 4 and 8 are very common cases that could be optimized. */
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for (; i < nb_samples; i++) {
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int j;
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int val = 0;
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int error_val = error_buffer[i];
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int error_sign;
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int d = *pred++;
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/* LPC prediction */
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for (j = 0; j < lpc_order; j++)
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val += (pred[j] - d) * lpc_coefs[j];
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val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
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val += d + error_val;
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buffer_out[i] = sign_extend(val, bps);
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/* adapt LPC coefficients */
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error_sign = sign_only(error_val);
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if (error_sign) {
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for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
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int sign;
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val = d - pred[j];
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sign = sign_only(val) * error_sign;
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lpc_coefs[j] -= sign;
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val *= sign;
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error_val -= (val >> lpc_quant) * (j + 1);
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}
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}
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}
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}
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static void decorrelate_stereo(int32_t *buffer[2], int nb_samples,
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int decorr_shift, int decorr_left_weight)
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{
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int i;
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for (i = 0; i < nb_samples; i++) {
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int32_t a, b;
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a = buffer[0][i];
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b = buffer[1][i];
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a -= (b * decorr_left_weight) >> decorr_shift;
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b += a;
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buffer[0][i] = b;
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buffer[1][i] = a;
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}
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}
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static void append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2],
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int extra_bits, int channels, int nb_samples)
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{
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int i, ch;
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for (ch = 0; ch < channels; ch++)
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for (i = 0; i < nb_samples; i++)
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buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
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}
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static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
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int channels)
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{
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ALACContext *alac = avctx->priv_data;
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int has_size, bps, is_compressed, decorr_shift, decorr_left_weight, ret;
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uint32_t output_samples;
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int i, ch;
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skip_bits(&alac->gb, 4); /* element instance tag */
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skip_bits(&alac->gb, 12); /* unused header bits */
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/* the number of output samples is stored in the frame */
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has_size = get_bits1(&alac->gb);
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alac->extra_bits = get_bits(&alac->gb, 2) << 3;
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bps = alac->sample_size - alac->extra_bits + channels - 1;
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if (bps > 32) {
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av_log(avctx, AV_LOG_ERROR, "bps is unsupported: %d\n", bps);
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return AVERROR_PATCHWELCOME;
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}
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/* whether the frame is compressed */
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is_compressed = !get_bits1(&alac->gb);
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if (has_size)
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output_samples = get_bits_long(&alac->gb, 32);
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else
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output_samples = alac->max_samples_per_frame;
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if (!output_samples || output_samples > alac->max_samples_per_frame) {
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av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %"PRIu32"\n",
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output_samples);
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return AVERROR_INVALIDDATA;
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}
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if (!alac->nb_samples) {
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/* get output buffer */
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frame->nb_samples = output_samples;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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} else if (output_samples != alac->nb_samples) {
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av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %"PRIu32" != %d\n",
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output_samples, alac->nb_samples);
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return AVERROR_INVALIDDATA;
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}
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alac->nb_samples = output_samples;
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if (alac->sample_size > 16) {
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for (ch = 0; ch < channels; ch++)
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alac->output_samples_buffer[ch] = (int32_t *)frame->extended_data[ch_index + ch];
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}
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if (is_compressed) {
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int16_t lpc_coefs[2][32];
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int lpc_order[2];
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int prediction_type[2];
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int lpc_quant[2];
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int rice_history_mult[2];
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if (!alac->rice_limit) {
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avpriv_request_sample(alac->avctx,
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"Compression with rice limit 0");
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return AVERROR(ENOSYS);
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}
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decorr_shift = get_bits(&alac->gb, 8);
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decorr_left_weight = get_bits(&alac->gb, 8);
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for (ch = 0; ch < channels; ch++) {
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prediction_type[ch] = get_bits(&alac->gb, 4);
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lpc_quant[ch] = get_bits(&alac->gb, 4);
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rice_history_mult[ch] = get_bits(&alac->gb, 3);
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lpc_order[ch] = get_bits(&alac->gb, 5);
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if (lpc_order[ch] >= alac->max_samples_per_frame)
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return AVERROR_INVALIDDATA;
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/* read the predictor table */
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for (i = lpc_order[ch] - 1; i >= 0; i--)
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lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);
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}
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if (alac->extra_bits) {
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for (i = 0; i < alac->nb_samples; i++) {
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for (ch = 0; ch < channels; ch++)
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alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
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}
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}
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for (ch = 0; ch < channels; ch++) {
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rice_decompress(alac, alac->predict_error_buffer[ch],
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alac->nb_samples, bps,
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rice_history_mult[ch] * alac->rice_history_mult / 4);
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/* adaptive FIR filter */
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if (prediction_type[ch] == 15) {
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/* Prediction type 15 runs the adaptive FIR twice.
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* The first pass uses the special-case coef_num = 31, while
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* the second pass uses the coefs from the bitstream.
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*
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* However, this prediction type is not currently used by the
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* reference encoder.
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*/
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lpc_prediction(alac->predict_error_buffer[ch],
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alac->predict_error_buffer[ch],
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alac->nb_samples, bps, NULL, 31, 0);
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} else if (prediction_type[ch] > 0) {
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av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
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prediction_type[ch]);
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}
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lpc_prediction(alac->predict_error_buffer[ch],
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alac->output_samples_buffer[ch], alac->nb_samples,
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bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
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}
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} else {
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/* not compressed, easy case */
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for (i = 0; i < alac->nb_samples; i++) {
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for (ch = 0; ch < channels; ch++) {
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alac->output_samples_buffer[ch][i] =
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get_sbits_long(&alac->gb, alac->sample_size);
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}
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}
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alac->extra_bits = 0;
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decorr_shift = 0;
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decorr_left_weight = 0;
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}
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if (channels == 2 && decorr_left_weight) {
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decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,
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decorr_shift, decorr_left_weight);
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}
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if (alac->extra_bits) {
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append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
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alac->extra_bits, channels, alac->nb_samples);
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}
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switch(alac->sample_size) {
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case 16: {
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for (ch = 0; ch < channels; ch++) {
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int16_t *outbuffer = (int16_t *)frame->extended_data[ch_index + ch];
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for (i = 0; i < alac->nb_samples; i++)
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*outbuffer++ = alac->output_samples_buffer[ch][i];
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}}
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break;
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case 24: {
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for (ch = 0; ch < channels; ch++) {
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for (i = 0; i < alac->nb_samples; i++)
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alac->output_samples_buffer[ch][i] <<= 8;
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}}
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break;
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}
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return 0;
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}
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static int alac_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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ALACContext *alac = avctx->priv_data;
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AVFrame *frame = data;
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enum AlacRawDataBlockType element;
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int channels;
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int ch, ret, got_end;
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init_get_bits(&alac->gb, avpkt->data, avpkt->size * 8);
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got_end = 0;
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alac->nb_samples = 0;
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ch = 0;
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while (get_bits_left(&alac->gb) >= 3) {
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element = get_bits(&alac->gb, 3);
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if (element == TYPE_END) {
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got_end = 1;
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break;
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}
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if (element > TYPE_CPE && element != TYPE_LFE) {
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av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d", element);
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return AVERROR_PATCHWELCOME;
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}
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channels = (element == TYPE_CPE) ? 2 : 1;
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if (ch + channels > alac->channels ||
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ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels) {
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av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
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return AVERROR_INVALIDDATA;
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}
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ret = decode_element(avctx, frame,
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ff_alac_channel_layout_offsets[alac->channels - 1][ch],
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channels);
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if (ret < 0 && get_bits_left(&alac->gb))
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return ret;
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ch += channels;
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}
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if (!got_end) {
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av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n");
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return AVERROR_INVALIDDATA;
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}
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if (!alac->nb_samples) {
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av_log(avctx, AV_LOG_ERROR, "No decodable data in the packet\n");
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return AVERROR_INVALIDDATA;
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}
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if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) {
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av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n",
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avpkt->size * 8 - get_bits_count(&alac->gb));
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}
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*got_frame_ptr = 1;
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return avpkt->size;
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}
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static av_cold int alac_decode_close(AVCodecContext *avctx)
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{
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ALACContext *alac = avctx->priv_data;
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int ch;
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for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
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av_freep(&alac->predict_error_buffer[ch]);
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if (alac->sample_size == 16)
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av_freep(&alac->output_samples_buffer[ch]);
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av_freep(&alac->extra_bits_buffer[ch]);
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}
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return 0;
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}
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static int allocate_buffers(ALACContext *alac)
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{
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int ch;
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int buf_size = alac->max_samples_per_frame * sizeof(int32_t);
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for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
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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,
|
|
};
|