597 lines
20 KiB
C
597 lines
20 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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* 32bit atom size
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* 32bit tag ("alac")
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* 32bit tag version (0)
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* 32bit samples per frame (used when not set explicitly in the frames)
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* 8bit compatible version (0)
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* 8bit sample size
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* 8bit history mult (40)
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* 8bit initial history (14)
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* 8bit rice param limit (10)
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* 8bit channels
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* 16bit maxRun (255)
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* 32bit max coded frame size (0 means unknown)
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* 32bit average bitrate (0 means unknown)
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* 32bit samplerate
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*/
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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 "unary.h"
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#include "mathops.h"
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#define ALAC_EXTRADATA_SIZE 36
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#define MAX_CHANNELS 2
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typedef struct {
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AVCodecContext *avctx;
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AVFrame frame;
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GetBitContext gb;
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int channels;
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/* buffers */
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int32_t *predict_error_buffer[MAX_CHANNELS];
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int32_t *output_samples_buffer[MAX_CHANNELS];
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int32_t *extra_bits_buffer[MAX_CHANNELS];
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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 int decode_scalar(GetBitContext *gb, int k, int readsamplesize)
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{
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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(gb, readsamplesize);
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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 bastardized_rice_decompress(ALACContext *alac,
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int32_t *output_buffer,
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int output_size,
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int readsamplesize,
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int rice_history_mult)
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{
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int output_count;
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unsigned int history = alac->rice_initial_history;
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int sign_modifier = 0;
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for (output_count = 0; output_count < output_size; output_count++) {
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int x, k;
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/* read k, that is bits as is */
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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, readsamplesize);
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x += sign_modifier;
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sign_modifier = 0;
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output_buffer[output_count] = (x >> 1) ^ -(x & 1);
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/* now 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) && (output_count+1 < output_size)) {
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int block_size;
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k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
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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 >= output_size - output_count){
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av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
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block_size= output_size - output_count - 1;
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}
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memset(&output_buffer[output_count + 1], 0,
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block_size * sizeof(*output_buffer));
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output_count += 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 predictor_decompress_fir_adapt(int32_t *error_buffer,
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int32_t *buffer_out,
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int output_size,
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int readsamplesize,
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int16_t *predictor_coef_table,
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int predictor_coef_num,
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int predictor_quantitization)
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{
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int i;
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/* first sample always copies */
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*buffer_out = *error_buffer;
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if (output_size <= 1)
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return;
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if (!predictor_coef_num) {
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memcpy(&buffer_out[1], &error_buffer[1],
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(output_size - 1) * sizeof(*buffer_out));
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return;
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}
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if (predictor_coef_num == 31) {
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/* simple 1st-order prediction */
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for (i = 1; i < output_size; i++) {
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buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],
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readsamplesize);
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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 = 0; i < predictor_coef_num; i++) {
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buffer_out[i + 1] = sign_extend(buffer_out[i] + error_buffer[i + 1],
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readsamplesize);
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}
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/* NOTE: 4 and 8 are very common cases that could be optimized. */
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/* general case */
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for (i = predictor_coef_num; i < output_size - 1; i++) {
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int j;
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int val = 0;
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int error_val = error_buffer[i + 1];
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int error_sign;
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int d = buffer_out[i - predictor_coef_num];
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for (j = 0; j < predictor_coef_num; j++) {
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val += (buffer_out[i - j] - d) *
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predictor_coef_table[j];
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}
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val = (val + (1 << (predictor_quantitization - 1))) >>
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predictor_quantitization;
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val += d + error_val;
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buffer_out[i + 1] = sign_extend(val, readsamplesize);
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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 = predictor_coef_num - 1; j >= 0 && error_val * error_sign > 0; j--) {
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int sign;
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val = d - buffer_out[i - j];
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sign = sign_only(val) * error_sign;
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predictor_coef_table[j] -= sign;
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val *= sign;
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error_val -= ((val >> predictor_quantitization) *
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(predictor_coef_num - j));
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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[MAX_CHANNELS],
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int numsamples, uint8_t interlacing_shift,
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uint8_t interlacing_leftweight)
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{
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int i;
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for (i = 0; i < numsamples; 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 * interlacing_leftweight) >> interlacing_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[MAX_CHANNELS],
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int32_t *extra_bits_buffer[MAX_CHANNELS],
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int extra_bits, int numchannels, int numsamples)
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{
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int i, ch;
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for (ch = 0; ch < numchannels; ch++)
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for (i = 0; i < numsamples; 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 void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS],
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int16_t *buffer_out, int numsamples)
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{
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int i;
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for (i = 0; i < numsamples; i++) {
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*buffer_out++ = buffer[0][i];
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*buffer_out++ = buffer[1][i];
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}
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}
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static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
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int32_t *buffer_out, int numsamples)
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{
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int i;
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for (i = 0; i < numsamples; i++) {
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*buffer_out++ = buffer[0][i] << 8;
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*buffer_out++ = buffer[1][i] << 8;
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}
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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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const uint8_t *inbuffer = avpkt->data;
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int input_buffer_size = avpkt->size;
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ALACContext *alac = avctx->priv_data;
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int channels;
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int hassize;
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unsigned int readsamplesize;
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int is_compressed;
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uint8_t interlacing_shift;
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uint8_t interlacing_leftweight;
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int i, ch, ret;
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init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
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channels = get_bits(&alac->gb, 3) + 1;
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if (channels != avctx->channels) {
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av_log(avctx, AV_LOG_ERROR, "frame header channel count mismatch\n");
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return AVERROR_INVALIDDATA;
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}
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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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hassize = get_bits1(&alac->gb);
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alac->extra_bits = get_bits(&alac->gb, 2) << 3;
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/* whether the frame is compressed */
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is_compressed = !get_bits1(&alac->gb);
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if (hassize) {
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/* now read the number of samples as a 32bit integer */
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uint32_t output_samples = get_bits_long(&alac->gb, 32);
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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: %d\n",
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output_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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} else
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alac->nb_samples = alac->max_samples_per_frame;
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/* get output buffer */
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alac->frame.nb_samples = alac->nb_samples;
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if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 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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readsamplesize = alac->sample_size - alac->extra_bits + channels - 1;
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if (readsamplesize > MIN_CACHE_BITS) {
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av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
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return -1;
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}
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if (is_compressed) {
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int16_t predictor_coef_table[MAX_CHANNELS][32];
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int predictor_coef_num[MAX_CHANNELS];
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int prediction_type[MAX_CHANNELS];
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int prediction_quantitization[MAX_CHANNELS];
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int ricemodifier[MAX_CHANNELS];
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interlacing_shift = get_bits(&alac->gb, 8);
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interlacing_leftweight = 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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prediction_quantitization[ch] = get_bits(&alac->gb, 4);
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ricemodifier[ch] = get_bits(&alac->gb, 3);
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predictor_coef_num[ch] = get_bits(&alac->gb, 5);
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/* read the predictor table */
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for (i = 0; i < predictor_coef_num[ch]; i++)
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predictor_coef_table[ch][i] = (int16_t)get_bits(&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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bastardized_rice_decompress(alac,
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alac->predict_error_buffer[ch],
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alac->nb_samples,
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readsamplesize,
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ricemodifier[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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predictor_decompress_fir_adapt(alac->predict_error_buffer[ch],
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alac->predict_error_buffer[ch],
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alac->nb_samples, readsamplesize,
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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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predictor_decompress_fir_adapt(alac->predict_error_buffer[ch],
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alac->output_samples_buffer[ch],
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alac->nb_samples, readsamplesize,
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predictor_coef_table[ch],
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predictor_coef_num[ch],
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prediction_quantitization[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] = get_sbits_long(&alac->gb,
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alac->sample_size);
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}
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}
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alac->extra_bits = 0;
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interlacing_shift = 0;
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interlacing_leftweight = 0;
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}
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if (get_bits(&alac->gb, 3) != 7)
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av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
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if (channels == 2 && interlacing_leftweight) {
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decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,
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interlacing_shift, interlacing_leftweight);
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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, alac->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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if (channels == 2) {
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interleave_stereo_16(alac->output_samples_buffer,
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(int16_t *)alac->frame.data[0],
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alac->nb_samples);
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} else {
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int16_t *outbuffer = (int16_t *)alac->frame.data[0];
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for (i = 0; i < alac->nb_samples; i++) {
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outbuffer[i] = alac->output_samples_buffer[0][i];
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}
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}
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break;
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case 24:
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if (channels == 2) {
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interleave_stereo_24(alac->output_samples_buffer,
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(int32_t *)alac->frame.data[0],
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alac->nb_samples);
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} else {
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int32_t *outbuffer = (int32_t *)alac->frame.data[0];
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for (i = 0; i < alac->nb_samples; i++)
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outbuffer[i] = alac->output_samples_buffer[0][i] << 8;
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}
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break;
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}
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if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
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av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
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*got_frame_ptr = 1;
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*(AVFrame *)data = alac->frame;
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return input_buffer_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 < alac->channels; ch++) {
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av_freep(&alac->predict_error_buffer[ch]);
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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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for (ch = 0; ch < alac->channels; ch++) {
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int buf_size = alac->max_samples_per_frame * sizeof(int32_t);
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FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],
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buf_size, buf_alloc_fail);
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FF_ALLOC_OR_GOTO(alac->avctx, alac->output_samples_buffer[ch],
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buf_size, buf_alloc_fail);
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FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
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buf_size, buf_alloc_fail);
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}
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return 0;
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buf_alloc_fail:
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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)"),
|
|
};
|