/* * 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- (0x24-)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: * bytes 0-3 atom size (0x24), big-endian * bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd) * bytes 8-35 data bytes needed by decoder * * Extradata: * 32bit size * 32bit tag (=alac) * 32bit zero? * 32bit max sample per frame * 8bit ?? (zero?) * 8bit sample size * 8bit history mult * 8bit initial history * 8bit kmodifier * 8bit channels? * 16bit ?? * 32bit max coded frame size * 32bit bitrate? * 32bit samplerate */ #include "avcodec.h" #include "get_bits.h" #include "bytestream.h" #include "unary.h" #include "mathops.h" #define ALAC_EXTRADATA_SIZE 36 #define MAX_CHANNELS 2 typedef struct { AVCodecContext *avctx; GetBitContext gb; int numchannels; int bytespersample; /* buffers */ int32_t *predicterror_buffer[MAX_CHANNELS]; int32_t *outputsamples_buffer[MAX_CHANNELS]; int32_t *wasted_bits_buffer[MAX_CHANNELS]; /* stuff from setinfo */ uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ uint8_t setinfo_sample_size; /* 0x10 */ uint8_t setinfo_rice_historymult; /* 0x28 */ uint8_t setinfo_rice_initialhistory; /* 0x0a */ uint8_t setinfo_rice_kmodifier; /* 0x0e */ /* end setinfo stuff */ int wasted_bits; } ALACContext; static void allocate_buffers(ALACContext *alac) { int chan; for (chan = 0; chan < MAX_CHANNELS; chan++) { alac->predicterror_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4); alac->outputsamples_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4); alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4); } } static int alac_set_info(ALACContext *alac) { const unsigned char *ptr = alac->avctx->extradata; ptr += 4; /* size */ ptr += 4; /* alac */ ptr += 4; /* 0 ? */ if(AV_RB32(ptr) >= UINT_MAX/4){ av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n"); return -1; } /* buffer size / 2 ? */ alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr); ptr++; /* ??? */ alac->setinfo_sample_size = *ptr++; if (alac->setinfo_sample_size > 32) { av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n"); return -1; } alac->setinfo_rice_historymult = *ptr++; alac->setinfo_rice_initialhistory = *ptr++; alac->setinfo_rice_kmodifier = *ptr++; ptr++; /* channels? */ bytestream_get_be16(&ptr); /* ??? */ bytestream_get_be32(&ptr); /* max coded frame size */ bytestream_get_be32(&ptr); /* bitrate ? */ bytestream_get_be32(&ptr); /* samplerate */ allocate_buffers(alac); return 0; } static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){ /* read x - number of 1s before 0 represent the rice */ int x = get_unary_0_9(gb); if (x > 8) { /* RICE THRESHOLD */ /* use alternative encoding */ x = get_bits(gb, readsamplesize); } else { if (k >= limit) k = limit; if (k != 1) { int extrabits = show_bits(gb, k); /* multiply x by 2^k - 1, as part of their strange algorithm */ x = (x << k) - x; if (extrabits > 1) { x += extrabits - 1; skip_bits(gb, k); } else skip_bits(gb, k - 1); } } return x; } static void bastardized_rice_decompress(ALACContext *alac, int32_t *output_buffer, int output_size, int readsamplesize, /* arg_10 */ int rice_initialhistory, /* arg424->b */ int rice_kmodifier, /* arg424->d */ int rice_historymult, /* arg424->c */ int rice_kmodifier_mask /* arg424->e */ ) { int output_count; unsigned int history = rice_initialhistory; int sign_modifier = 0; for (output_count = 0; output_count < output_size; output_count++) { int32_t x; int32_t x_modified; int32_t final_val; /* standard rice encoding */ int k; /* size of extra bits */ /* read k, that is bits as is */ k = av_log2((history >> 9) + 3); x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize); x_modified = sign_modifier + x; final_val = (x_modified + 1) / 2; if (x_modified & 1) final_val *= -1; output_buffer[output_count] = final_val; sign_modifier = 0; /* now update the history */ history += x_modified * rice_historymult - ((history * rice_historymult) >> 9); if (x_modified > 0xffff) history = 0xffff; /* special case: there may be compressed blocks of 0 */ if ((history < 128) && (output_count+1 < output_size)) { int k; unsigned int block_size; sign_modifier = 1; k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */); block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16); if (block_size > 0) { if(block_size >= output_size - output_count){ av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count); block_size= output_size - output_count - 1; } memset(&output_buffer[output_count+1], 0, block_size * 4); output_count += block_size; } if (block_size > 0xffff) sign_modifier = 0; history = 0; } } } static inline int sign_only(int v) { return v ? FFSIGN(v) : 0; } static void predictor_decompress_fir_adapt(int32_t *error_buffer, int32_t *buffer_out, int output_size, int readsamplesize, int16_t *predictor_coef_table, int predictor_coef_num, int predictor_quantitization) { int i; /* first sample always copies */ *buffer_out = *error_buffer; if (!predictor_coef_num) { if (output_size <= 1) return; memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4); return; } if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */ /* second-best case scenario for fir decompression, * error describes a small difference from the previous sample only */ if (output_size <= 1) return; for (i = 0; i < output_size - 1; i++) { int32_t prev_value; int32_t error_value; prev_value = buffer_out[i]; error_value = error_buffer[i+1]; buffer_out[i+1] = sign_extend((prev_value + error_value), readsamplesize); } return; } /* read warm-up samples */ if (predictor_coef_num > 0) for (i = 0; i < predictor_coef_num; i++) { int32_t val; val = buffer_out[i] + error_buffer[i+1]; val = sign_extend(val, readsamplesize); buffer_out[i+1] = val; } #if 0 /* 4 and 8 are very common cases (the only ones i've seen). these * should be unrolled and optimized */ if (predictor_coef_num == 4) { /* FIXME: optimized general case */ return; } if (predictor_coef_table == 8) { /* FIXME: optimized general case */ return; } #endif /* general case */ if (predictor_coef_num > 0) { for (i = predictor_coef_num + 1; i < output_size; i++) { int j; int sum = 0; int outval; int error_val = error_buffer[i]; for (j = 0; j < predictor_coef_num; j++) { sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) * predictor_coef_table[j]; } outval = (1 << (predictor_quantitization-1)) + sum; outval = outval >> predictor_quantitization; outval = outval + buffer_out[0] + error_val; outval = sign_extend(outval, readsamplesize); buffer_out[predictor_coef_num+1] = outval; if (error_val > 0) { int predictor_num = predictor_coef_num - 1; while (predictor_num >= 0 && error_val > 0) { int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; int sign = sign_only(val); predictor_coef_table[predictor_num] -= sign; val *= sign; /* absolute value */ error_val -= ((val >> predictor_quantitization) * (predictor_coef_num - predictor_num)); predictor_num--; } } else if (error_val < 0) { int predictor_num = predictor_coef_num - 1; while (predictor_num >= 0 && error_val < 0) { int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; int sign = - sign_only(val); predictor_coef_table[predictor_num] -= sign; val *= sign; /* neg value */ error_val -= ((val >> predictor_quantitization) * (predictor_coef_num - predictor_num)); predictor_num--; } } buffer_out++; } } } static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS], int16_t *buffer_out, int numchannels, int numsamples, uint8_t interlacing_shift, uint8_t interlacing_leftweight) { int i; if (numsamples <= 0) return; /* weighted interlacing */ if (interlacing_leftweight) { for (i = 0; i < numsamples; i++) { int32_t a, b; a = buffer[0][i]; b = buffer[1][i]; a -= (b * interlacing_leftweight) >> interlacing_shift; b += a; buffer_out[i*numchannels] = b; buffer_out[i*numchannels + 1] = a; } return; } /* otherwise basic interlacing took place */ for (i = 0; i < numsamples; i++) { int16_t left, right; left = buffer[0][i]; right = buffer[1][i]; buffer_out[i*numchannels] = left; buffer_out[i*numchannels + 1] = right; } } static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS], int32_t *buffer_out, int32_t *wasted_bits_buffer[MAX_CHANNELS], int wasted_bits, int numchannels, int numsamples, uint8_t interlacing_shift, uint8_t interlacing_leftweight) { int i; if (numsamples <= 0) return; /* weighted interlacing */ if (interlacing_leftweight) { for (i = 0; i < numsamples; i++) { int32_t a, b; a = buffer[0][i]; b = buffer[1][i]; a -= (b * interlacing_leftweight) >> interlacing_shift; b += a; if (wasted_bits) { b = (b << wasted_bits) | wasted_bits_buffer[0][i]; a = (a << wasted_bits) | wasted_bits_buffer[1][i]; } buffer_out[i * numchannels] = b << 8; buffer_out[i * numchannels + 1] = a << 8; } } else { for (i = 0; i < numsamples; i++) { int32_t left, right; left = buffer[0][i]; right = buffer[1][i]; if (wasted_bits) { left = (left << wasted_bits) | wasted_bits_buffer[0][i]; right = (right << wasted_bits) | wasted_bits_buffer[1][i]; } buffer_out[i * numchannels] = left << 8; buffer_out[i * numchannels + 1] = right << 8; } } } static int alac_decode_frame(AVCodecContext *avctx, void *outbuffer, int *outputsize, AVPacket *avpkt) { const uint8_t *inbuffer = avpkt->data; int input_buffer_size = avpkt->size; ALACContext *alac = avctx->priv_data; int channels; unsigned int outputsamples; int hassize; unsigned int readsamplesize; int isnotcompressed; uint8_t interlacing_shift; uint8_t interlacing_leftweight; /* short-circuit null buffers */ if (!inbuffer || !input_buffer_size) return -1; init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); channels = get_bits(&alac->gb, 3) + 1; if (channels > MAX_CHANNELS) { av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n", MAX_CHANNELS); return -1; } /* 2^result = something to do with output waiting. * perhaps matters if we read > 1 frame in a pass? */ skip_bits(&alac->gb, 4); skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */ /* the output sample size is stored soon */ hassize = get_bits1(&alac->gb); alac->wasted_bits = get_bits(&alac->gb, 2) << 3; /* whether the frame is compressed */ isnotcompressed = get_bits1(&alac->gb); if (hassize) { /* now read the number of samples as a 32bit integer */ outputsamples = get_bits_long(&alac->gb, 32); if(outputsamples > alac->setinfo_max_samples_per_frame){ av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame); return -1; } } else outputsamples = alac->setinfo_max_samples_per_frame; switch (alac->setinfo_sample_size) { case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16; alac->bytespersample = channels << 1; break; case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32; alac->bytespersample = channels << 2; break; default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n", alac->setinfo_sample_size); return -1; } if(outputsamples > *outputsize / alac->bytespersample){ av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n"); return -1; } *outputsize = outputsamples * alac->bytespersample; readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1; if (readsamplesize > MIN_CACHE_BITS) { av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize); return -1; } if (!isnotcompressed) { /* so it is compressed */ int16_t predictor_coef_table[MAX_CHANNELS][32]; int predictor_coef_num[MAX_CHANNELS]; int prediction_type[MAX_CHANNELS]; int prediction_quantitization[MAX_CHANNELS]; int ricemodifier[MAX_CHANNELS]; int i, chan; interlacing_shift = get_bits(&alac->gb, 8); interlacing_leftweight = get_bits(&alac->gb, 8); for (chan = 0; chan < channels; chan++) { prediction_type[chan] = get_bits(&alac->gb, 4); prediction_quantitization[chan] = get_bits(&alac->gb, 4); ricemodifier[chan] = get_bits(&alac->gb, 3); predictor_coef_num[chan] = get_bits(&alac->gb, 5); /* read the predictor table */ for (i = 0; i < predictor_coef_num[chan]; i++) predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); } if (alac->wasted_bits) { int i, ch; for (i = 0; i < outputsamples; i++) { for (ch = 0; ch < channels; ch++) alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits); } } for (chan = 0; chan < channels; chan++) { bastardized_rice_decompress(alac, alac->predicterror_buffer[chan], outputsamples, readsamplesize, alac->setinfo_rice_initialhistory, alac->setinfo_rice_kmodifier, ricemodifier[chan] * alac->setinfo_rice_historymult / 4, (1 << alac->setinfo_rice_kmodifier) - 1); if (prediction_type[chan] == 0) { /* adaptive fir */ predictor_decompress_fir_adapt(alac->predicterror_buffer[chan], alac->outputsamples_buffer[chan], outputsamples, readsamplesize, predictor_coef_table[chan], predictor_coef_num[chan], prediction_quantitization[chan]); } else { av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]); /* I think the only other prediction type (or perhaps this is * just a boolean?) runs adaptive fir twice.. like: * predictor_decompress_fir_adapt(predictor_error, tempout, ...) * predictor_decompress_fir_adapt(predictor_error, outputsamples ...) * little strange.. */ } } } else { /* not compressed, easy case */ int i, chan; if (alac->setinfo_sample_size <= 16) { for (i = 0; i < outputsamples; i++) for (chan = 0; chan < channels; chan++) { int32_t audiobits; audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size); alac->outputsamples_buffer[chan][i] = audiobits; } } else { for (i = 0; i < outputsamples; i++) { for (chan = 0; chan < channels; chan++) { alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb, alac->setinfo_sample_size); alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i], alac->setinfo_sample_size); } } } alac->wasted_bits = 0; interlacing_shift = 0; interlacing_leftweight = 0; } if (get_bits(&alac->gb, 3) != 7) av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n"); switch(alac->setinfo_sample_size) { case 16: if (channels == 2) { reconstruct_stereo_16(alac->outputsamples_buffer, (int16_t*)outbuffer, alac->numchannels, outputsamples, interlacing_shift, interlacing_leftweight); } else { int i; for (i = 0; i < outputsamples; i++) { ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i]; } } break; case 24: if (channels == 2) { decorrelate_stereo_24(alac->outputsamples_buffer, outbuffer, alac->wasted_bits_buffer, alac->wasted_bits, alac->numchannels, outputsamples, interlacing_shift, interlacing_leftweight); } else { int i; for (i = 0; i < outputsamples; i++) ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8; } break; } if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8) av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb)); return input_buffer_size; } static av_cold int alac_decode_init(AVCodecContext * avctx) { ALACContext *alac = avctx->priv_data; alac->avctx = avctx; alac->numchannels = alac->avctx->channels; /* 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; } return 0; } static av_cold int alac_decode_close(AVCodecContext *avctx) { ALACContext *alac = avctx->priv_data; int chan; for (chan = 0; chan < MAX_CHANNELS; chan++) { av_freep(&alac->predicterror_buffer[chan]); av_freep(&alac->outputsamples_buffer[chan]); av_freep(&alac->wasted_bits_buffer[chan]); } return 0; } AVCodec ff_alac_decoder = { "alac", AVMEDIA_TYPE_AUDIO, CODEC_ID_ALAC, sizeof(ALACContext), alac_decode_init, NULL, alac_decode_close, alac_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), };