Mercurial > libavcodec.hg
view mlpdec.c @ 7451:85ab7655ad4d libavcodec
Modify all codecs to report their supported input and output sample format(s).
| author | pross |
|---|---|
| date | Thu, 31 Jul 2008 10:47:31 +0000 |
| parents | e3822c61f2e4 |
| children | 88ffd7c9c0ed |
line wrap: on
line source
/* * MLP decoder * Copyright (c) 2007-2008 Ian Caulfield * * 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 mlpdec.c * MLP decoder */ #include <stdint.h> #include "avcodec.h" #include "libavutil/intreadwrite.h" #include "bitstream.h" #include "libavutil/crc.h" #include "parser.h" #include "mlp_parser.h" /** Maximum number of channels that can be decoded. */ #define MAX_CHANNELS 16 /** Maximum number of matrices used in decoding; most streams have one matrix * per output channel, but some rematrix a channel (usually 0) more than once. */ #define MAX_MATRICES 15 /** Maximum number of substreams that can be decoded. This could also be set * higher, but I haven't seen any examples with more than two. */ #define MAX_SUBSTREAMS 2 /** maximum sample frequency seen in files */ #define MAX_SAMPLERATE 192000 /** maximum number of audio samples within one access unit */ #define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000)) /** next power of two greater than MAX_BLOCKSIZE */ #define MAX_BLOCKSIZE_POW2 (64 * (MAX_SAMPLERATE / 48000)) /** number of allowed filters */ #define NUM_FILTERS 2 /** The maximum number of taps in either the IIR or FIR filter; * I believe MLP actually specifies the maximum order for IIR filters as four, * and that the sum of the orders of both filters must be <= 8. */ #define MAX_FILTER_ORDER 8 /** number of bits used for VLC lookup - longest Huffman code is 9 */ #define VLC_BITS 9 static const char* sample_message = "Please file a bug report following the instructions at " "http://ffmpeg.mplayerhq.hu/bugreports.html and include " "a sample of this file."; typedef struct SubStream { //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded. uint8_t restart_seen; //@{ /** restart header data */ //! The type of noise to be used in the rematrix stage. uint16_t noise_type; //! The index of the first channel coded in this substream. uint8_t min_channel; //! The index of the last channel coded in this substream. uint8_t max_channel; //! The number of channels input into the rematrix stage. uint8_t max_matrix_channel; //! The left shift applied to random noise in 0x31ea substreams. uint8_t noise_shift; //! The current seed value for the pseudorandom noise generator(s). uint32_t noisegen_seed; //! Set if the substream contains extra info to check the size of VLC blocks. uint8_t data_check_present; //! Bitmask of which parameter sets are conveyed in a decoding parameter block. uint8_t param_presence_flags; #define PARAM_BLOCKSIZE (1 << 7) #define PARAM_MATRIX (1 << 6) #define PARAM_OUTSHIFT (1 << 5) #define PARAM_QUANTSTEP (1 << 4) #define PARAM_FIR (1 << 3) #define PARAM_IIR (1 << 2) #define PARAM_HUFFOFFSET (1 << 1) //@} //@{ /** matrix data */ //! Number of matrices to be applied. uint8_t num_primitive_matrices; //! matrix output channel uint8_t matrix_out_ch[MAX_MATRICES]; //! Whether the LSBs of the matrix output are encoded in the bitstream. uint8_t lsb_bypass[MAX_MATRICES]; //! Matrix coefficients, stored as 2.14 fixed point. int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2]; //! Left shift to apply to noise values in 0x31eb substreams. uint8_t matrix_noise_shift[MAX_MATRICES]; //@} //! Left shift to apply to Huffman-decoded residuals. uint8_t quant_step_size[MAX_CHANNELS]; //! number of PCM samples in current audio block uint16_t blocksize; //! Number of PCM samples decoded so far in this frame. uint16_t blockpos; //! Left shift to apply to decoded PCM values to get final 24-bit output. int8_t output_shift[MAX_CHANNELS]; //! Running XOR of all output samples. int32_t lossless_check_data; } SubStream; typedef struct MLPDecodeContext { AVCodecContext *avctx; //! Set if a valid major sync block has been read. Otherwise no decoding is possible. uint8_t params_valid; //! Number of substreams contained within this stream. uint8_t num_substreams; //! Index of the last substream to decode - further substreams are skipped. uint8_t max_decoded_substream; //! number of PCM samples contained in each frame int access_unit_size; //! next power of two above the number of samples in each frame int access_unit_size_pow2; SubStream substream[MAX_SUBSTREAMS]; //@{ /** filter data */ #define FIR 0 #define IIR 1 //! number of taps in filter uint8_t filter_order[MAX_CHANNELS][NUM_FILTERS]; //! Right shift to apply to output of filter. uint8_t filter_shift[MAX_CHANNELS][NUM_FILTERS]; int32_t filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER]; int32_t filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER]; //@} //@{ /** sample data coding information */ //! Offset to apply to residual values. int16_t huff_offset[MAX_CHANNELS]; //! sign/rounding-corrected version of huff_offset int32_t sign_huff_offset[MAX_CHANNELS]; //! Which VLC codebook to use to read residuals. uint8_t codebook[MAX_CHANNELS]; //! Size of residual suffix not encoded using VLC. uint8_t huff_lsbs[MAX_CHANNELS]; //@} int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2]; } MLPDecodeContext; /** Tables defining the Huffman codes. * There are three entropy coding methods used in MLP (four if you count * "none" as a method). These use the same sequences for codes starting with * 00 or 01, but have different codes starting with 1. */ static const uint8_t huffman_tables[3][18][2] = { { /* Huffman table 0, -7 - +10 */ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3}, {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, }, { /* Huffman table 1, -7 - +8 */ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, {0x02, 2}, {0x03, 2}, {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, }, { /* Huffman table 2, -7 - +7 */ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, {0x01, 1}, {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, } }; static VLC huff_vlc[3]; static int crc_init = 0; static AVCRC crc_63[1024]; static AVCRC crc_1D[1024]; /** Initialize static data, constant between all invocations of the codec. */ static av_cold void init_static() { INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, &huffman_tables[0][0][1], 2, 1, &huffman_tables[0][0][0], 2, 1, 512); INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, &huffman_tables[1][0][1], 2, 1, &huffman_tables[1][0][0], 2, 1, 512); INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, &huffman_tables[2][0][1], 2, 1, &huffman_tables[2][0][0], 2, 1, 512); if (!crc_init) { av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63)); av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D)); crc_init = 1; } } /** MLP uses checksums that seem to be based on the standard CRC algorithm, but * are not (in implementation terms, the table lookup and XOR are reversed). * We can implement this behavior using a standard av_crc on all but the * last element, then XOR that with the last element. */ static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size) { uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c checksum ^= buf[buf_size-1]; return checksum; } /** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8 * number of bits, starting two bits into the first byte of buf. */ static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size) { int i; int num_bytes = (bit_size + 2) / 8; int crc = crc_1D[buf[0] & 0x3f]; crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2); crc ^= buf[num_bytes - 1]; for (i = 0; i < ((bit_size + 2) & 7); i++) { crc <<= 1; if (crc & 0x100) crc ^= 0x11D; crc ^= (buf[num_bytes] >> (7 - i)) & 1; } return crc; } static inline int32_t calculate_sign_huff(MLPDecodeContext *m, unsigned int substr, unsigned int ch) { SubStream *s = &m->substream[substr]; int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch]; int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1); int32_t sign_huff_offset = m->huff_offset[ch]; if (m->codebook[ch] > 0) sign_huff_offset -= 7 << lsb_bits; if (sign_shift >= 0) sign_huff_offset -= 1 << sign_shift; return sign_huff_offset; } /** Read a sample, consisting of either, both or neither of entropy-coded MSBs * and plain LSBs. */ static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr, unsigned int pos) { SubStream *s = &m->substream[substr]; unsigned int mat, channel; for (mat = 0; mat < s->num_primitive_matrices; mat++) if (s->lsb_bypass[mat]) m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp); for (channel = s->min_channel; channel <= s->max_channel; channel++) { int codebook = m->codebook[channel]; int quant_step_size = s->quant_step_size[channel]; int lsb_bits = m->huff_lsbs[channel] - quant_step_size; int result = 0; if (codebook > 0) result = get_vlc2(gbp, huff_vlc[codebook-1].table, VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); if (result < 0) return -1; if (lsb_bits > 0) result = (result << lsb_bits) + get_bits(gbp, lsb_bits); result += m->sign_huff_offset[channel]; result <<= quant_step_size; m->sample_buffer[pos + s->blockpos][channel] = result; } return 0; } static av_cold int mlp_decode_init(AVCodecContext *avctx) { MLPDecodeContext *m = avctx->priv_data; int substr; init_static(); m->avctx = avctx; for (substr = 0; substr < MAX_SUBSTREAMS; substr++) m->substream[substr].lossless_check_data = 0xffffffff; avctx->sample_fmt = SAMPLE_FMT_S16; return 0; } /** Read a major sync info header - contains high level information about * the stream - sample rate, channel arrangement etc. Most of this * information is not actually necessary for decoding, only for playback. */ static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb) { MLPHeaderInfo mh; int substr; if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0) return -1; if (mh.group1_bits == 0) { av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n"); return -1; } if (mh.group2_bits > mh.group1_bits) { av_log(m->avctx, AV_LOG_ERROR, "Channel group 2 cannot have more bits per sample than group 1.\n"); return -1; } if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) { av_log(m->avctx, AV_LOG_ERROR, "Channel groups with differing sample rates are not currently supported.\n"); return -1; } if (mh.group1_samplerate == 0) { av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n"); return -1; } if (mh.group1_samplerate > MAX_SAMPLERATE) { av_log(m->avctx, AV_LOG_ERROR, "Sampling rate %d is greater than the supported maximum (%d).\n", mh.group1_samplerate, MAX_SAMPLERATE); return -1; } if (mh.access_unit_size > MAX_BLOCKSIZE) { av_log(m->avctx, AV_LOG_ERROR, "Block size %d is greater than the supported maximum (%d).\n", mh.access_unit_size, MAX_BLOCKSIZE); return -1; } if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) { av_log(m->avctx, AV_LOG_ERROR, "Block size pow2 %d is greater than the supported maximum (%d).\n", mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2); return -1; } if (mh.num_substreams == 0) return -1; if (mh.num_substreams > MAX_SUBSTREAMS) { av_log(m->avctx, AV_LOG_ERROR, "Number of substreams %d is larger than the maximum supported " "by the decoder. %s\n", mh.num_substreams, sample_message); return -1; } m->access_unit_size = mh.access_unit_size; m->access_unit_size_pow2 = mh.access_unit_size_pow2; m->num_substreams = mh.num_substreams; m->max_decoded_substream = m->num_substreams - 1; m->avctx->sample_rate = mh.group1_samplerate; m->avctx->frame_size = mh.access_unit_size; #ifdef CONFIG_AUDIO_NONSHORT m->avctx->bits_per_sample = mh.group1_bits; if (mh.group1_bits > 16) { m->avctx->sample_fmt = SAMPLE_FMT_S32; } #endif m->params_valid = 1; for (substr = 0; substr < MAX_SUBSTREAMS; substr++) m->substream[substr].restart_seen = 0; return 0; } /** Read a restart header from a block in a substream. This contains parameters * required to decode the audio that do not change very often. Generally * (always) present only in blocks following a major sync. */ static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, const uint8_t *buf, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int ch; int sync_word, tmp; uint8_t checksum; uint8_t lossless_check; int start_count = get_bits_count(gbp); sync_word = get_bits(gbp, 13); if (sync_word != 0x31ea >> 1) { av_log(m->avctx, AV_LOG_ERROR, "restart header sync incorrect (got 0x%04x)\n", sync_word); return -1; } s->noise_type = get_bits1(gbp); skip_bits(gbp, 16); /* Output timestamp */ s->min_channel = get_bits(gbp, 4); s->max_channel = get_bits(gbp, 4); s->max_matrix_channel = get_bits(gbp, 4); if (s->min_channel > s->max_channel) { av_log(m->avctx, AV_LOG_ERROR, "Substream min channel cannot be greater than max channel.\n"); return -1; } if (m->avctx->request_channels > 0 && s->max_channel + 1 >= m->avctx->request_channels && substr < m->max_decoded_substream) { av_log(m->avctx, AV_LOG_INFO, "Extracting %d channel downmix from substream %d. " "Further substreams will be skipped.\n", s->max_channel + 1, substr); m->max_decoded_substream = substr; } s->noise_shift = get_bits(gbp, 4); s->noisegen_seed = get_bits(gbp, 23); skip_bits(gbp, 19); s->data_check_present = get_bits1(gbp); lossless_check = get_bits(gbp, 8); if (substr == m->max_decoded_substream && s->lossless_check_data != 0xffffffff) { tmp = s->lossless_check_data; tmp ^= tmp >> 16; tmp ^= tmp >> 8; tmp &= 0xff; if (tmp != lossless_check) av_log(m->avctx, AV_LOG_WARNING, "Lossless check failed - expected %02x, calculated %02x.\n", lossless_check, tmp); else dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n", substr, tmp); } skip_bits(gbp, 16); for (ch = 0; ch <= s->max_matrix_channel; ch++) { int ch_assign = get_bits(gbp, 6); dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch, ch_assign); if (ch_assign != ch) { av_log(m->avctx, AV_LOG_ERROR, "Non-1:1 channel assignments are used in this stream. %s\n", sample_message); return -1; } } checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count); if (checksum != get_bits(gbp, 8)) av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n"); /* Set default decoding parameters. */ s->param_presence_flags = 0xff; s->num_primitive_matrices = 0; s->blocksize = 8; s->lossless_check_data = 0; memset(s->output_shift , 0, sizeof(s->output_shift )); memset(s->quant_step_size, 0, sizeof(s->quant_step_size)); for (ch = s->min_channel; ch <= s->max_channel; ch++) { m->filter_order[ch][FIR] = 0; m->filter_order[ch][IIR] = 0; m->filter_shift[ch][FIR] = 0; m->filter_shift[ch][IIR] = 0; /* Default audio coding is 24-bit raw PCM. */ m->huff_offset [ch] = 0; m->sign_huff_offset[ch] = (-1) << 23; m->codebook [ch] = 0; m->huff_lsbs [ch] = 24; } if (substr == m->max_decoded_substream) { m->avctx->channels = s->max_channel + 1; } return 0; } /** Read parameters for one of the prediction filters. */ static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, unsigned int channel, unsigned int filter) { const char fchar = filter ? 'I' : 'F'; int i, order; // Filter is 0 for FIR, 1 for IIR. assert(filter < 2); order = get_bits(gbp, 4); if (order > MAX_FILTER_ORDER) { av_log(m->avctx, AV_LOG_ERROR, "%cIR filter order %d is greater than maximum %d.\n", fchar, order, MAX_FILTER_ORDER); return -1; } m->filter_order[channel][filter] = order; if (order > 0) { int coeff_bits, coeff_shift; m->filter_shift[channel][filter] = get_bits(gbp, 4); coeff_bits = get_bits(gbp, 5); coeff_shift = get_bits(gbp, 3); if (coeff_bits < 1 || coeff_bits > 16) { av_log(m->avctx, AV_LOG_ERROR, "%cIR filter coeff_bits must be between 1 and 16.\n", fchar); return -1; } if (coeff_bits + coeff_shift > 16) { av_log(m->avctx, AV_LOG_ERROR, "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n", fchar); return -1; } for (i = 0; i < order; i++) m->filter_coeff[channel][filter][i] = get_sbits(gbp, coeff_bits) << coeff_shift; if (get_bits1(gbp)) { int state_bits, state_shift; if (filter == FIR) { av_log(m->avctx, AV_LOG_ERROR, "FIR filter has state data specified.\n"); return -1; } state_bits = get_bits(gbp, 4); state_shift = get_bits(gbp, 4); /* TODO: Check validity of state data. */ for (i = 0; i < order; i++) m->filter_state[channel][filter][i] = get_sbits(gbp, state_bits) << state_shift; } } return 0; } /** Read decoding parameters that change more often than those in the restart * header. */ static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int mat, ch; if (get_bits1(gbp)) s->param_presence_flags = get_bits(gbp, 8); if (s->param_presence_flags & PARAM_BLOCKSIZE) if (get_bits1(gbp)) { s->blocksize = get_bits(gbp, 9); if (s->blocksize > MAX_BLOCKSIZE) { av_log(m->avctx, AV_LOG_ERROR, "block size too large\n"); s->blocksize = 0; return -1; } } if (s->param_presence_flags & PARAM_MATRIX) if (get_bits1(gbp)) { s->num_primitive_matrices = get_bits(gbp, 4); for (mat = 0; mat < s->num_primitive_matrices; mat++) { int frac_bits, max_chan; s->matrix_out_ch[mat] = get_bits(gbp, 4); frac_bits = get_bits(gbp, 4); s->lsb_bypass [mat] = get_bits1(gbp); if (s->matrix_out_ch[mat] > s->max_channel) { av_log(m->avctx, AV_LOG_ERROR, "Invalid channel %d specified as output from matrix.\n", s->matrix_out_ch[mat]); return -1; } if (frac_bits > 14) { av_log(m->avctx, AV_LOG_ERROR, "Too many fractional bits specified.\n"); return -1; } max_chan = s->max_matrix_channel; if (!s->noise_type) max_chan+=2; for (ch = 0; ch <= max_chan; ch++) { int coeff_val = 0; if (get_bits1(gbp)) coeff_val = get_sbits(gbp, frac_bits + 2); s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); } if (s->noise_type) s->matrix_noise_shift[mat] = get_bits(gbp, 4); else s->matrix_noise_shift[mat] = 0; } } if (s->param_presence_flags & PARAM_OUTSHIFT) if (get_bits1(gbp)) for (ch = 0; ch <= s->max_matrix_channel; ch++) { s->output_shift[ch] = get_bits(gbp, 4); dprintf(m->avctx, "output shift[%d] = %d\n", ch, s->output_shift[ch]); /* TODO: validate */ } if (s->param_presence_flags & PARAM_QUANTSTEP) if (get_bits1(gbp)) for (ch = 0; ch <= s->max_channel; ch++) { s->quant_step_size[ch] = get_bits(gbp, 4); /* TODO: validate */ m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch); } for (ch = s->min_channel; ch <= s->max_channel; ch++) if (get_bits1(gbp)) { if (s->param_presence_flags & PARAM_FIR) if (get_bits1(gbp)) if (read_filter_params(m, gbp, ch, FIR) < 0) return -1; if (s->param_presence_flags & PARAM_IIR) if (get_bits1(gbp)) if (read_filter_params(m, gbp, ch, IIR) < 0) return -1; if (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] && m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) { av_log(m->avctx, AV_LOG_ERROR, "FIR and IIR filters must use the same precision.\n"); return -1; } /* The FIR and IIR filters must have the same precision. * To simplify the filtering code, only the precision of the * FIR filter is considered. If only the IIR filter is employed, * the FIR filter precision is set to that of the IIR filter, so * that the filtering code can use it. */ if (!m->filter_order[ch][FIR] && m->filter_order[ch][IIR]) m->filter_shift[ch][FIR] = m->filter_shift[ch][IIR]; if (s->param_presence_flags & PARAM_HUFFOFFSET) if (get_bits1(gbp)) m->huff_offset[ch] = get_sbits(gbp, 15); m->codebook [ch] = get_bits(gbp, 2); m->huff_lsbs[ch] = get_bits(gbp, 5); m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch); /* TODO: validate */ } return 0; } #define MSB_MASK(bits) (-1u << bits) /** Generate PCM samples using the prediction filters and residual values * read from the data stream, and update the filter state. */ static void filter_channel(MLPDecodeContext *m, unsigned int substr, unsigned int channel) { SubStream *s = &m->substream[substr]; int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER]; unsigned int filter_shift = m->filter_shift[channel][FIR]; int32_t mask = MSB_MASK(s->quant_step_size[channel]); int index = MAX_BLOCKSIZE; int j, i; for (j = 0; j < NUM_FILTERS; j++) { memcpy(& filter_state_buffer [j][MAX_BLOCKSIZE], &m->filter_state[channel][j][0], MAX_FILTER_ORDER * sizeof(int32_t)); } for (i = 0; i < s->blocksize; i++) { int32_t residual = m->sample_buffer[i + s->blockpos][channel]; unsigned int order; int64_t accum = 0; int32_t result; /* TODO: Move this code to DSPContext? */ for (j = 0; j < NUM_FILTERS; j++) for (order = 0; order < m->filter_order[channel][j]; order++) accum += (int64_t)filter_state_buffer[j][index + order] * m->filter_coeff[channel][j][order]; accum = accum >> filter_shift; result = (accum + residual) & mask; --index; filter_state_buffer[FIR][index] = result; filter_state_buffer[IIR][index] = result - accum; m->sample_buffer[i + s->blockpos][channel] = result; } for (j = 0; j < NUM_FILTERS; j++) { memcpy(&m->filter_state[channel][j][0], & filter_state_buffer [j][index], MAX_FILTER_ORDER * sizeof(int32_t)); } } /** Read a block of PCM residual data (or actual if no filtering active). */ static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int i, ch, expected_stream_pos = 0; if (s->data_check_present) { expected_stream_pos = get_bits_count(gbp); expected_stream_pos += get_bits(gbp, 16); av_log(m->avctx, AV_LOG_WARNING, "This file contains some features " "we have not tested yet. %s\n", sample_message); } if (s->blockpos + s->blocksize > m->access_unit_size) { av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n"); return -1; } memset(&m->bypassed_lsbs[s->blockpos][0], 0, s->blocksize * sizeof(m->bypassed_lsbs[0])); for (i = 0; i < s->blocksize; i++) { if (read_huff_channels(m, gbp, substr, i) < 0) return -1; } for (ch = s->min_channel; ch <= s->max_channel; ch++) { filter_channel(m, substr, ch); } s->blockpos += s->blocksize; if (s->data_check_present) { if (get_bits_count(gbp) != expected_stream_pos) av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n"); skip_bits(gbp, 8); } return 0; } /** Data table used for TrueHD noise generation function. */ static const int8_t noise_table[256] = { 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2, 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62, 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5, 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40, 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34, 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30, 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36, 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69, 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24, 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20, 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23, 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8, 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40, 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37, 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52, -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70, }; /** Noise generation functions. * I'm not sure what these are for - they seem to be some kind of pseudorandom * sequence generators, used to generate noise data which is used when the * channels are rematrixed. I'm not sure if they provide a practical benefit * to compression, or just obfuscate the decoder. Are they for some kind of * dithering? */ /** Generate two channels of noise, used in the matrix when * restart sync word == 0x31ea. */ static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int i; uint32_t seed = s->noisegen_seed; unsigned int maxchan = s->max_matrix_channel; for (i = 0; i < s->blockpos; i++) { uint16_t seed_shr7 = seed >> 7; m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift; m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift; seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5); } s->noisegen_seed = seed; } /** Generate a block of noise, used when restart sync word == 0x31eb. */ static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int i; uint32_t seed = s->noisegen_seed; for (i = 0; i < m->access_unit_size_pow2; i++) { uint8_t seed_shr15 = seed >> 15; m->noise_buffer[i] = noise_table[seed_shr15]; seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5); } s->noisegen_seed = seed; } /** Apply the channel matrices in turn to reconstruct the original audio * samples. */ static void rematrix_channels(MLPDecodeContext *m, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int mat, src_ch, i; unsigned int maxchan; maxchan = s->max_matrix_channel; if (!s->noise_type) { generate_2_noise_channels(m, substr); maxchan += 2; } else { fill_noise_buffer(m, substr); } for (mat = 0; mat < s->num_primitive_matrices; mat++) { int matrix_noise_shift = s->matrix_noise_shift[mat]; unsigned int dest_ch = s->matrix_out_ch[mat]; int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]); /* TODO: DSPContext? */ for (i = 0; i < s->blockpos; i++) { int64_t accum = 0; for (src_ch = 0; src_ch <= maxchan; src_ch++) { accum += (int64_t)m->sample_buffer[i][src_ch] * s->matrix_coeff[mat][src_ch]; } if (matrix_noise_shift) { uint32_t index = s->num_primitive_matrices - mat; index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1); accum += m->noise_buffer[index] << (matrix_noise_shift + 7); } m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask) + m->bypassed_lsbs[i][mat]; } } } /** Write the audio data into the output buffer. */ static int output_data_internal(MLPDecodeContext *m, unsigned int substr, uint8_t *data, unsigned int *data_size, int is32) { SubStream *s = &m->substream[substr]; unsigned int i, ch = 0; int32_t *data_32 = (int32_t*) data; int16_t *data_16 = (int16_t*) data; if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2)) return -1; for (i = 0; i < s->blockpos; i++) { for (ch = 0; ch <= s->max_channel; ch++) { int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch]; s->lossless_check_data ^= (sample & 0xffffff) << ch; if (is32) *data_32++ = sample << 8; else *data_16++ = sample >> 8; } } *data_size = i * ch * (is32 ? 4 : 2); return 0; } static int output_data(MLPDecodeContext *m, unsigned int substr, uint8_t *data, unsigned int *data_size) { if (m->avctx->sample_fmt == SAMPLE_FMT_S32) return output_data_internal(m, substr, data, data_size, 1); else return output_data_internal(m, substr, data, data_size, 0); } /** XOR together all the bytes of a buffer. * Does this belong in dspcontext? */ static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size) { uint32_t scratch = 0; const uint8_t *buf_end = buf + buf_size; for (; buf < buf_end - 3; buf += 4) scratch ^= *((const uint32_t*)buf); scratch ^= scratch >> 16; scratch ^= scratch >> 8; for (; buf < buf_end; buf++) scratch ^= *buf; return scratch; } /** Read an access unit from the stream. * Returns < 0 on error, 0 if not enough data is present in the input stream * otherwise returns the number of bytes consumed. */ static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, const uint8_t *buf, int buf_size) { MLPDecodeContext *m = avctx->priv_data; GetBitContext gb; unsigned int length, substr; unsigned int substream_start; unsigned int header_size = 4; unsigned int substr_header_size = 0; uint8_t substream_parity_present[MAX_SUBSTREAMS]; uint16_t substream_data_len[MAX_SUBSTREAMS]; uint8_t parity_bits; if (buf_size < 4) return 0; length = (AV_RB16(buf) & 0xfff) * 2; if (length > buf_size) return -1; init_get_bits(&gb, (buf + 4), (length - 4) * 8); if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) { dprintf(m->avctx, "Found major sync.\n"); if (read_major_sync(m, &gb) < 0) goto error; header_size += 28; } if (!m->params_valid) { av_log(m->avctx, AV_LOG_WARNING, "Stream parameters not seen; skipping frame.\n"); *data_size = 0; return length; } substream_start = 0; for (substr = 0; substr < m->num_substreams; substr++) { int extraword_present, checkdata_present, end; extraword_present = get_bits1(&gb); skip_bits1(&gb); checkdata_present = get_bits1(&gb); skip_bits1(&gb); end = get_bits(&gb, 12) * 2; substr_header_size += 2; if (extraword_present) { skip_bits(&gb, 16); substr_header_size += 2; } if (end + header_size + substr_header_size > length) { av_log(m->avctx, AV_LOG_ERROR, "Indicated length of substream %d data goes off end of " "packet.\n", substr); end = length - header_size - substr_header_size; } if (end < substream_start) { av_log(avctx, AV_LOG_ERROR, "Indicated end offset of substream %d data " "is smaller than calculated start offset.\n", substr); goto error; } if (substr > m->max_decoded_substream) continue; substream_parity_present[substr] = checkdata_present; substream_data_len[substr] = end - substream_start; substream_start = end; } parity_bits = calculate_parity(buf, 4); parity_bits ^= calculate_parity(buf + header_size, substr_header_size); if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) { av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n"); goto error; } buf += header_size + substr_header_size; for (substr = 0; substr <= m->max_decoded_substream; substr++) { SubStream *s = &m->substream[substr]; init_get_bits(&gb, buf, substream_data_len[substr] * 8); s->blockpos = 0; do { if (get_bits1(&gb)) { if (get_bits1(&gb)) { /* A restart header should be present. */ if (read_restart_header(m, &gb, buf, substr) < 0) goto next_substr; s->restart_seen = 1; } if (!s->restart_seen) { av_log(m->avctx, AV_LOG_ERROR, "No restart header present in substream %d.\n", substr); goto next_substr; } if (read_decoding_params(m, &gb, substr) < 0) goto next_substr; } if (!s->restart_seen) { av_log(m->avctx, AV_LOG_ERROR, "No restart header present in substream %d.\n", substr); goto next_substr; } if (read_block_data(m, &gb, substr) < 0) return -1; } while ((get_bits_count(&gb) < substream_data_len[substr] * 8) && get_bits1(&gb) == 0); skip_bits(&gb, (-get_bits_count(&gb)) & 15); if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 && (show_bits_long(&gb, 32) == 0xd234d234 || show_bits_long(&gb, 20) == 0xd234e)) { skip_bits(&gb, 18); if (substr == m->max_decoded_substream) av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n"); if (get_bits1(&gb)) { int shorten_by = get_bits(&gb, 13); shorten_by = FFMIN(shorten_by, s->blockpos); s->blockpos -= shorten_by; } else skip_bits(&gb, 13); } if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 && substream_parity_present[substr]) { uint8_t parity, checksum; parity = calculate_parity(buf, substream_data_len[substr] - 2); if ((parity ^ get_bits(&gb, 8)) != 0xa9) av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr); checksum = mlp_checksum8(buf, substream_data_len[substr] - 2); if (checksum != get_bits(&gb, 8)) av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n", substr); } if (substream_data_len[substr] * 8 != get_bits_count(&gb)) { av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr); return -1; } next_substr: buf += substream_data_len[substr]; } rematrix_channels(m, m->max_decoded_substream); if (output_data(m, m->max_decoded_substream, data, data_size) < 0) return -1; return length; error: m->params_valid = 0; return -1; } AVCodec mlp_decoder = { "mlp", CODEC_TYPE_AUDIO, CODEC_ID_MLP, sizeof(MLPDecodeContext), mlp_decode_init, NULL, NULL, read_access_unit, .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"), };