/* * ALAC (Apple Lossless Audio Codec) decoder * Copyright (c) 2005 David Hammerton * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * ALAC (Apple Lossless Audio Codec) decoder * @author 2005 David Hammerton * @see http://crazney.net/programs/itunes/alac.html * * Note: This decoder expects a 36-byte QuickTime atom to be * passed through the extradata[_size] fields. This atom is tacked onto * the end of an 'alac' stsd atom and has the following format: * * 32bit atom size * 32bit tag ("alac") * 32bit tag version (0) * 32bit samples per frame (used when not set explicitly in the frames) * 8bit compatible version (0) * 8bit sample size * 8bit history mult (40) * 8bit initial history (10) * 8bit rice param limit (14) * 8bit channels * 16bit maxRun (255) * 32bit max coded frame size (0 means unknown) * 32bit average bitrate (0 means unknown) * 32bit samplerate */ #include <inttypes.h> #include "libavutil/channel_layout.h" #include "avcodec.h" #include "get_bits.h" #include "bytestream.h" #include "internal.h" #include "thread.h" #include "unary.h" #include "mathops.h" #include "alac_data.h" #define ALAC_EXTRADATA_SIZE 36 typedef struct { AVCodecContext *avctx; GetBitContext gb; int channels; int32_t *predict_error_buffer[2]; int32_t *output_samples_buffer[2]; int32_t *extra_bits_buffer[2]; uint32_t max_samples_per_frame; uint8_t sample_size; uint8_t rice_history_mult; uint8_t rice_initial_history; uint8_t rice_limit; int extra_bits; /**< number of extra bits beyond 16-bit */ int nb_samples; /**< number of samples in the current frame */ int direct_output; } ALACContext; static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps) { unsigned int x = get_unary_0_9(gb); if (x > 8) { /* RICE THRESHOLD */ /* use alternative encoding */ x = get_bits_long(gb, bps); } else if (k != 1) { int extrabits = show_bits(gb, k); /* multiply x by 2^k - 1, as part of their strange algorithm */ x = (x << k) - x; if (extrabits > 1) { x += extrabits - 1; skip_bits(gb, k); } else skip_bits(gb, k - 1); } return x; } static int rice_decompress(ALACContext *alac, int32_t *output_buffer, int nb_samples, int bps, int rice_history_mult) { int i; unsigned int history = alac->rice_initial_history; int sign_modifier = 0; for (i = 0; i < nb_samples; i++) { int k; unsigned int x; if(get_bits_left(&alac->gb) <= 0) return -1; /* calculate rice param and decode next value */ k = av_log2((history >> 9) + 3); k = FFMIN(k, alac->rice_limit); x = decode_scalar(&alac->gb, k, bps); x += sign_modifier; sign_modifier = 0; output_buffer[i] = (x >> 1) ^ -(x & 1); /* update the history */ if (x > 0xffff) history = 0xffff; else history += x * rice_history_mult - ((history * rice_history_mult) >> 9); /* special case: there may be compressed blocks of 0 */ if ((history < 128) && (i + 1 < nb_samples)) { int block_size; /* calculate rice param and decode block size */ k = 7 - av_log2(history) + ((history + 16) >> 6); k = FFMIN(k, alac->rice_limit); block_size = decode_scalar(&alac->gb, k, 16); if (block_size > 0) { if (block_size >= nb_samples - i) { av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, nb_samples, i); block_size = nb_samples - i - 1; } memset(&output_buffer[i + 1], 0, block_size * sizeof(*output_buffer)); i += block_size; } if (block_size <= 0xffff) sign_modifier = 1; history = 0; } } return 0; } static inline int sign_only(int v) { return v ? FFSIGN(v) : 0; } static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out, int nb_samples, int bps, int16_t *lpc_coefs, int lpc_order, int lpc_quant) { int i; int32_t *pred = buffer_out; /* first sample always copies */ *buffer_out = *error_buffer; if (nb_samples <= 1) return; if (!lpc_order) { memcpy(&buffer_out[1], &error_buffer[1], (nb_samples - 1) * sizeof(*buffer_out)); return; } if (lpc_order == 31) { /* simple 1st-order prediction */ for (i = 1; i < nb_samples; i++) { buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps); } return; } /* read warm-up samples */ for (i = 1; i <= lpc_order && i < nb_samples; i++) buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps); /* NOTE: 4 and 8 are very common cases that could be optimized. */ for (; i < nb_samples; i++) { int j; int val = 0; int error_val = error_buffer[i]; int error_sign; int d = *pred++; /* LPC prediction */ for (j = 0; j < lpc_order; j++) val += (pred[j] - d) * lpc_coefs[j]; val = (val + (1 << (lpc_quant - 1))) >> lpc_quant; val += d + error_val; buffer_out[i] = sign_extend(val, bps); /* adapt LPC coefficients */ error_sign = sign_only(error_val); if (error_sign) { for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) { int sign; val = d - pred[j]; sign = sign_only(val) * error_sign; lpc_coefs[j] -= sign; val *= sign; error_val -= (val >> lpc_quant) * (j + 1); } } } } static void decorrelate_stereo(int32_t *buffer[2], int nb_samples, int decorr_shift, int decorr_left_weight) { int i; for (i = 0; i < nb_samples; i++) { int32_t a, b; a = buffer[0][i]; b = buffer[1][i]; a -= (b * decorr_left_weight) >> decorr_shift; b += a; buffer[0][i] = b; buffer[1][i] = a; } } static void append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2], int extra_bits, int channels, int nb_samples) { int i, ch; for (ch = 0; ch < channels; ch++) for (i = 0; i < nb_samples; i++) buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i]; } static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, int channels) { ALACContext *alac = avctx->priv_data; int has_size, bps, is_compressed, decorr_shift, decorr_left_weight, ret; uint32_t output_samples; int i, ch; skip_bits(&alac->gb, 4); /* element instance tag */ skip_bits(&alac->gb, 12); /* unused header bits */ /* the number of output samples is stored in the frame */ has_size = get_bits1(&alac->gb); alac->extra_bits = get_bits(&alac->gb, 2) << 3; bps = alac->sample_size - alac->extra_bits + channels - 1; if (bps > 32U) { av_log(avctx, AV_LOG_ERROR, "bps is unsupported: %d\n", bps); return AVERROR_PATCHWELCOME; } /* whether the frame is compressed */ is_compressed = !get_bits1(&alac->gb); if (has_size) output_samples = get_bits_long(&alac->gb, 32); else output_samples = alac->max_samples_per_frame; if (!output_samples || output_samples > alac->max_samples_per_frame) { av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %"PRIu32"\n", output_samples); return AVERROR_INVALIDDATA; } if (!alac->nb_samples) { ThreadFrame tframe = { .f = frame }; /* get output buffer */ frame->nb_samples = output_samples; if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0) return ret; } else if (output_samples != alac->nb_samples) { av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %"PRIu32" != %d\n", output_samples, alac->nb_samples); return AVERROR_INVALIDDATA; } alac->nb_samples = output_samples; if (alac->direct_output) { for (ch = 0; ch < channels; ch++) alac->output_samples_buffer[ch] = (int32_t *)frame->extended_data[ch_index + ch]; } if (is_compressed) { int16_t lpc_coefs[2][32]; int lpc_order[2]; int prediction_type[2]; int lpc_quant[2]; int rice_history_mult[2]; decorr_shift = get_bits(&alac->gb, 8); decorr_left_weight = get_bits(&alac->gb, 8); for (ch = 0; ch < channels; ch++) { prediction_type[ch] = get_bits(&alac->gb, 4); lpc_quant[ch] = get_bits(&alac->gb, 4); rice_history_mult[ch] = get_bits(&alac->gb, 3); lpc_order[ch] = get_bits(&alac->gb, 5); if (lpc_order[ch] >= alac->max_samples_per_frame) return AVERROR_INVALIDDATA; /* read the predictor table */ for (i = lpc_order[ch] - 1; i >= 0; i--) lpc_coefs[ch][i] = get_sbits(&alac->gb, 16); } if (alac->extra_bits) { for (i = 0; i < alac->nb_samples; i++) { if(get_bits_left(&alac->gb) <= 0) return -1; for (ch = 0; ch < channels; ch++) alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits); } } for (ch = 0; ch < channels; ch++) { int ret=rice_decompress(alac, alac->predict_error_buffer[ch], alac->nb_samples, bps, rice_history_mult[ch] * alac->rice_history_mult / 4); if(ret<0) return ret; /* adaptive FIR filter */ if (prediction_type[ch] == 15) { /* Prediction type 15 runs the adaptive FIR twice. * The first pass uses the special-case coef_num = 31, while * the second pass uses the coefs from the bitstream. * * However, this prediction type is not currently used by the * reference encoder. */ lpc_prediction(alac->predict_error_buffer[ch], alac->predict_error_buffer[ch], alac->nb_samples, bps, NULL, 31, 0); } else if (prediction_type[ch] > 0) { av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n", prediction_type[ch]); } lpc_prediction(alac->predict_error_buffer[ch], alac->output_samples_buffer[ch], alac->nb_samples, bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]); } } else { /* not compressed, easy case */ for (i = 0; i < alac->nb_samples; i++) { if(get_bits_left(&alac->gb) <= 0) return -1; for (ch = 0; ch < channels; ch++) { alac->output_samples_buffer[ch][i] = get_sbits_long(&alac->gb, alac->sample_size); } } alac->extra_bits = 0; decorr_shift = 0; decorr_left_weight = 0; } if (channels == 2 && decorr_left_weight) { decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples, decorr_shift, decorr_left_weight); } if (alac->extra_bits) { append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer, alac->extra_bits, channels, alac->nb_samples); } if(av_sample_fmt_is_planar(avctx->sample_fmt)) { switch(alac->sample_size) { case 16: { for (ch = 0; ch < channels; ch++) { int16_t *outbuffer = (int16_t *)frame->extended_data[ch_index + ch]; for (i = 0; i < alac->nb_samples; i++) *outbuffer++ = alac->output_samples_buffer[ch][i]; }} break; case 24: { for (ch = 0; ch < channels; ch++) { for (i = 0; i < alac->nb_samples; i++) alac->output_samples_buffer[ch][i] <<= 8; }} break; } }else{ switch(alac->sample_size) { case 16: { int16_t *outbuffer = ((int16_t *)frame->extended_data[0]) + ch_index; for (i = 0; i < alac->nb_samples; i++) { for (ch = 0; ch < channels; ch++) *outbuffer++ = alac->output_samples_buffer[ch][i]; outbuffer += alac->channels - channels; } } break; case 24: { int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index; for (i = 0; i < alac->nb_samples; i++) { for (ch = 0; ch < channels; ch++) *outbuffer++ = alac->output_samples_buffer[ch][i] << 8; outbuffer += alac->channels - channels; } } break; case 32: { int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index; for (i = 0; i < alac->nb_samples; i++) { for (ch = 0; ch < channels; ch++) *outbuffer++ = alac->output_samples_buffer[ch][i]; outbuffer += alac->channels - channels; } } break; } } return 0; } static int alac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { ALACContext *alac = avctx->priv_data; AVFrame *frame = data; enum AlacRawDataBlockType element; int channels; int ch, ret, got_end; if ((ret = init_get_bits8(&alac->gb, avpkt->data, avpkt->size)) < 0) return ret; got_end = 0; alac->nb_samples = 0; ch = 0; while (get_bits_left(&alac->gb) >= 3) { element = get_bits(&alac->gb, 3); if (element == TYPE_END) { got_end = 1; break; } if (element > TYPE_CPE && element != TYPE_LFE) { av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d\n", element); return AVERROR_PATCHWELCOME; } channels = (element == TYPE_CPE) ? 2 : 1; if (ch + channels > alac->channels || ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels) { av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n"); return AVERROR_INVALIDDATA; } ret = decode_element(avctx, frame, ff_alac_channel_layout_offsets[alac->channels - 1][ch], channels); if (ret < 0 && get_bits_left(&alac->gb)) return ret; ch += channels; } if (!got_end) { av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n"); return AVERROR_INVALIDDATA; } if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) { av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", avpkt->size * 8 - get_bits_count(&alac->gb)); } if (alac->channels == ch) *got_frame_ptr = 1; else av_log(avctx, AV_LOG_WARNING, "Failed to decode all channels\n"); return avpkt->size; } static av_cold int alac_decode_close(AVCodecContext *avctx) { ALACContext *alac = avctx->priv_data; int ch; for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) { av_freep(&alac->predict_error_buffer[ch]); if (!alac->direct_output) av_freep(&alac->output_samples_buffer[ch]); av_freep(&alac->extra_bits_buffer[ch]); } return 0; } static int allocate_buffers(ALACContext *alac) { int ch; int buf_size = alac->max_samples_per_frame * sizeof(int32_t); for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) { FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch], buf_size, buf_alloc_fail); alac->direct_output = alac->sample_size > 16 && av_sample_fmt_is_planar(alac->avctx->sample_fmt); if (!alac->direct_output) { FF_ALLOC_OR_GOTO(alac->avctx, alac->output_samples_buffer[ch], buf_size, buf_alloc_fail); } FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch], buf_size, buf_alloc_fail); } return 0; buf_alloc_fail: alac_decode_close(alac->avctx); return AVERROR(ENOMEM); } static int alac_set_info(ALACContext *alac) { GetByteContext gb; bytestream2_init(&gb, alac->avctx->extradata, alac->avctx->extradata_size); bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4 alac->max_samples_per_frame = bytestream2_get_be32u(&gb); if (!alac->max_samples_per_frame || alac->max_samples_per_frame > INT_MAX / sizeof(int32_t)) { av_log(alac->avctx, AV_LOG_ERROR, "max samples per frame invalid: %"PRIu32"\n", alac->max_samples_per_frame); return AVERROR_INVALIDDATA; } bytestream2_skipu(&gb, 1); // compatible version alac->sample_size = bytestream2_get_byteu(&gb); alac->rice_history_mult = bytestream2_get_byteu(&gb); alac->rice_initial_history = bytestream2_get_byteu(&gb); alac->rice_limit = bytestream2_get_byteu(&gb); alac->channels = bytestream2_get_byteu(&gb); bytestream2_get_be16u(&gb); // maxRun bytestream2_get_be32u(&gb); // max coded frame size bytestream2_get_be32u(&gb); // average bitrate bytestream2_get_be32u(&gb); // samplerate return 0; } static av_cold int alac_decode_init(AVCodecContext * avctx) { int ret; int req_packed; ALACContext *alac = avctx->priv_data; alac->avctx = avctx; /* initialize from the extradata */ if (alac->avctx->extradata_size < ALAC_EXTRADATA_SIZE) { av_log(avctx, AV_LOG_ERROR, "extradata is too small\n"); return AVERROR_INVALIDDATA; } if (alac_set_info(alac)) { av_log(avctx, AV_LOG_ERROR, "set_info failed\n"); return -1; } req_packed = LIBAVCODEC_VERSION_MAJOR < 55 && !av_sample_fmt_is_planar(avctx->request_sample_fmt); switch (alac->sample_size) { case 16: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_S16P; break; case 24: case 32: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S32P; break; default: avpriv_request_sample(avctx, "Sample depth %d", alac->sample_size); return AVERROR_PATCHWELCOME; } avctx->bits_per_raw_sample = alac->sample_size; if (alac->channels < 1) { av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n"); alac->channels = avctx->channels; } else { if (alac->channels > ALAC_MAX_CHANNELS) alac->channels = avctx->channels; else avctx->channels = alac->channels; } if (avctx->channels > ALAC_MAX_CHANNELS || avctx->channels <= 0 ) { av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n", avctx->channels); return AVERROR_PATCHWELCOME; } avctx->channel_layout = ff_alac_channel_layouts[alac->channels - 1]; if ((ret = allocate_buffers(alac)) < 0) { av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n"); return ret; } return 0; } static int init_thread_copy(AVCodecContext *avctx) { ALACContext *alac = avctx->priv_data; alac->avctx = avctx; return allocate_buffers(alac); } AVCodec ff_alac_decoder = { .name = "alac", .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_ALAC, .priv_data_size = sizeof(ALACContext), .init = alac_decode_init, .close = alac_decode_close, .decode = alac_decode_frame, .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy), .capabilities = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS, };