/* * MLP decoder * Copyright (c) 2007-2008 Ian Caulfield * * This file is part of Libav. * * Libav 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. * * Libav 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 Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * MLP decoder */ #include #include "libavutil/internal.h" #include "libavutil/intreadwrite.h" #include "libavutil/channel_layout.h" #include "libavutil/crc.h" #include "avcodec.h" #include "bitstream.h" #include "internal.h" #include "parser.h" #include "mlp_parser.h" #include "mlpdsp.h" #include "mlp.h" #include "config.h" /** number of bits used for VLC lookup - longest Huffman code is 9 */ #if ARCH_ARM #define VLC_BITS 5 #define VLC_STATIC_SIZE 64 #else #define VLC_BITS 9 #define VLC_STATIC_SIZE 512 #endif 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; /// For each channel output by the matrix, the output channel to map it to uint8_t ch_assign[MAX_CHANNELS]; /// The channel layout for this substream uint64_t mask; /// The matrix encoding mode for this substream enum AVMatrixEncoding matrix_encoding; /// Channel coding parameters for channels in the substream ChannelParams channel_params[MAX_CHANNELS]; /// 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) #define PARAM_PRESENCE (1 << 0) //@} //@{ /** 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]; /// 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; /// Current access unit being read has a major sync. int is_major_sync_unit; /// Size of the major sync unit, in bytes int major_sync_header_size; /// 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]; int matrix_changed; int filter_changed[MAX_CHANNELS][NUM_FILTERS]; int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS]; MLPDSPContext dsp; } MLPDecodeContext; static const uint64_t thd_channel_order[] = { AV_CH_FRONT_LEFT, AV_CH_FRONT_RIGHT, // LR AV_CH_FRONT_CENTER, // C AV_CH_LOW_FREQUENCY, // LFE AV_CH_SIDE_LEFT, AV_CH_SIDE_RIGHT, // LRs AV_CH_TOP_FRONT_LEFT, AV_CH_TOP_FRONT_RIGHT, // LRvh AV_CH_FRONT_LEFT_OF_CENTER, AV_CH_FRONT_RIGHT_OF_CENTER, // LRc AV_CH_BACK_LEFT, AV_CH_BACK_RIGHT, // LRrs AV_CH_BACK_CENTER, // Cs AV_CH_TOP_CENTER, // Ts AV_CH_SURROUND_DIRECT_LEFT, AV_CH_SURROUND_DIRECT_RIGHT, // LRsd AV_CH_WIDE_LEFT, AV_CH_WIDE_RIGHT, // LRw AV_CH_TOP_FRONT_CENTER, // Cvh AV_CH_LOW_FREQUENCY_2, // LFE2 }; static int mlp_channel_layout_subset(uint64_t channel_layout, uint64_t mask) { return channel_layout && ((channel_layout & mask) == channel_layout); } static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout, int index) { int i; if (av_get_channel_layout_nb_channels(channel_layout) <= index) return 0; for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++) if (channel_layout & thd_channel_order[i] && !index--) return thd_channel_order[i]; return 0; } static VLC huff_vlc[3]; /** Initialize static data, constant between all invocations of the codec. */ static av_cold void init_static(void) { if (!huff_vlc[0].bits) { INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, &ff_mlp_huffman_tables[0][0][1], 2, 1, &ff_mlp_huffman_tables[0][0][0], 2, 1, VLC_STATIC_SIZE); INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, &ff_mlp_huffman_tables[1][0][1], 2, 1, &ff_mlp_huffman_tables[1][0][0], 2, 1, VLC_STATIC_SIZE); INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, &ff_mlp_huffman_tables[2][0][1], 2, 1, &ff_mlp_huffman_tables[2][0][0], 2, 1, VLC_STATIC_SIZE); } ff_mlp_init_crc(); } static inline int32_t calculate_sign_huff(MLPDecodeContext *m, unsigned int substr, unsigned int ch) { SubStream *s = &m->substream[substr]; ChannelParams *cp = &s->channel_params[ch]; int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch]; int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1); int32_t sign_huff_offset = cp->huff_offset; if (cp->codebook > 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, BitstreamContext *bc, 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] = bitstream_read_bit(bc); for (channel = s->min_channel; channel <= s->max_channel; channel++) { ChannelParams *cp = &s->channel_params[channel]; int codebook = cp->codebook; int quant_step_size = s->quant_step_size[channel]; int lsb_bits = cp->huff_lsbs - quant_step_size; int result = 0; if (codebook > 0) result = bitstream_read_vlc(bc, huff_vlc[codebook-1].table, VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); if (result < 0) return AVERROR_INVALIDDATA; if (lsb_bits > 0) result = (result << lsb_bits) + bitstream_read(bc, lsb_bits); result += cp->sign_huff_offset; 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; ff_mlpdsp_init(&m->dsp); 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, BitstreamContext *bc) { MLPHeaderInfo mh; int substr, ret; if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, bc)) != 0) return ret; if (mh.group1_bits == 0) { av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n"); return AVERROR_INVALIDDATA; } 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 AVERROR_INVALIDDATA; } 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 AVERROR_INVALIDDATA; } if (mh.group1_samplerate == 0) { av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n"); return AVERROR_INVALIDDATA; } 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 AVERROR_INVALIDDATA; } 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 AVERROR_INVALIDDATA; } 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 AVERROR_INVALIDDATA; } if (mh.num_substreams == 0) return AVERROR_INVALIDDATA; if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) { av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n"); return AVERROR_INVALIDDATA; } if (mh.num_substreams > MAX_SUBSTREAMS) { avpriv_request_sample(m->avctx, "%d substreams (more than the " "maximum supported by the decoder)", mh.num_substreams); return AVERROR_PATCHWELCOME; } m->major_sync_header_size = mh.header_size; m->access_unit_size = mh.access_unit_size; m->access_unit_size_pow2 = mh.access_unit_size_pow2; m->num_substreams = mh.num_substreams; /* limit to decoding 3 substreams, as the 4th is used by Dolby Atmos for non-audio data */ m->max_decoded_substream = FFMIN(m->num_substreams - 1, 2); m->avctx->sample_rate = mh.group1_samplerate; m->avctx->frame_size = mh.access_unit_size; m->avctx->bits_per_raw_sample = mh.group1_bits; if (mh.group1_bits > 16) m->avctx->sample_fmt = AV_SAMPLE_FMT_S32; else m->avctx->sample_fmt = AV_SAMPLE_FMT_S16; m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(m->substream[m->max_decoded_substream].ch_assign, m->substream[m->max_decoded_substream].output_shift, m->substream[m->max_decoded_substream].max_matrix_channel, m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); m->params_valid = 1; for (substr = 0; substr < MAX_SUBSTREAMS; substr++) m->substream[substr].restart_seen = 0; /* Set the layout for each substream. When there's more than one, the first * substream is Stereo. Subsequent substreams' layouts are indicated in the * major sync. */ if (m->avctx->codec_id == AV_CODEC_ID_MLP) { if ((substr = (mh.num_substreams > 1))) m->substream[0].mask = AV_CH_LAYOUT_STEREO; m->substream[substr].mask = mh.channel_layout_mlp; } else { if ((substr = (mh.num_substreams > 1))) m->substream[0].mask = AV_CH_LAYOUT_STEREO; if (mh.num_substreams > 2) if (mh.channel_layout_thd_stream2) m->substream[2].mask = mh.channel_layout_thd_stream2; else m->substream[2].mask = mh.channel_layout_thd_stream1; m->substream[substr].mask = mh.channel_layout_thd_stream1; } /* Parse the TrueHD decoder channel modifiers and set each substream's * AVMatrixEncoding accordingly. * * The meaning of the modifiers depends on the channel layout: * * - THD_CH_MODIFIER_LTRT, THD_CH_MODIFIER_LBINRBIN only apply to 2-channel * * - THD_CH_MODIFIER_MONO applies to 1-channel or 2-channel (dual mono) * * - THD_CH_MODIFIER_SURROUNDEX, THD_CH_MODIFIER_NOTSURROUNDEX only apply to * layouts with an Ls/Rs channel pair */ for (substr = 0; substr < MAX_SUBSTREAMS; substr++) m->substream[substr].matrix_encoding = AV_MATRIX_ENCODING_NONE; if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) { if (mh.num_substreams > 2 && mh.channel_layout_thd_stream2 & AV_CH_SIDE_LEFT && mh.channel_layout_thd_stream2 & AV_CH_SIDE_RIGHT && mh.channel_modifier_thd_stream2 == THD_CH_MODIFIER_SURROUNDEX) m->substream[2].matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX; if (mh.num_substreams > 1 && mh.channel_layout_thd_stream1 & AV_CH_SIDE_LEFT && mh.channel_layout_thd_stream1 & AV_CH_SIDE_RIGHT && mh.channel_modifier_thd_stream1 == THD_CH_MODIFIER_SURROUNDEX) m->substream[1].matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX; if (mh.num_substreams > 0) switch (mh.channel_modifier_thd_stream0) { case THD_CH_MODIFIER_LTRT: m->substream[0].matrix_encoding = AV_MATRIX_ENCODING_DOLBY; break; case THD_CH_MODIFIER_LBINRBIN: m->substream[0].matrix_encoding = AV_MATRIX_ENCODING_DOLBYHEADPHONE; break; default: break; } } 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, BitstreamContext *bc, 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 = bitstream_tell(bc); int min_channel, max_channel, max_matrix_channel; const int std_max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP ? MAX_MATRIX_CHANNEL_MLP : MAX_MATRIX_CHANNEL_TRUEHD; sync_word = bitstream_read(bc, 13); if (sync_word != 0x31ea >> 1) { av_log(m->avctx, AV_LOG_ERROR, "restart header sync incorrect (got 0x%04x)\n", sync_word); return AVERROR_INVALIDDATA; } s->noise_type = bitstream_read_bit(bc); if (m->avctx->codec_id == AV_CODEC_ID_MLP && s->noise_type) { av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n"); return AVERROR_INVALIDDATA; } bitstream_skip(bc, 16); /* Output timestamp */ min_channel = bitstream_read(bc, 4); max_channel = bitstream_read(bc, 4); max_matrix_channel = bitstream_read(bc, 4); if (max_matrix_channel > std_max_matrix_channel) { av_log(m->avctx, AV_LOG_ERROR, "Max matrix channel cannot be greater than %d.\n", max_matrix_channel); return AVERROR_INVALIDDATA; } if (max_channel != max_matrix_channel) { av_log(m->avctx, AV_LOG_ERROR, "Max channel must be equal max matrix channel.\n"); return AVERROR_INVALIDDATA; } /* This should happen for TrueHD streams with >6 channels and MLP's noise * type. It is not yet known if this is allowed. */ if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) { avpriv_request_sample(m->avctx, "%d channels (more than the " "maximum supported by the decoder)", s->max_channel + 2); return AVERROR_PATCHWELCOME; } if (min_channel > max_channel) { av_log(m->avctx, AV_LOG_ERROR, "Substream min channel cannot be greater than max channel.\n"); return AVERROR_INVALIDDATA; } s->min_channel = min_channel; s->max_channel = max_channel; s->max_matrix_channel = max_matrix_channel; if (mlp_channel_layout_subset(m->avctx->request_channel_layout, s->mask) && m->max_decoded_substream > substr) { av_log(m->avctx, AV_LOG_DEBUG, "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. " "Further substreams will be skipped.\n", s->max_channel + 1, s->mask, substr); m->max_decoded_substream = substr; } s->noise_shift = bitstream_read(bc, 4); s->noisegen_seed = bitstream_read(bc, 23); bitstream_skip(bc, 19); s->data_check_present = bitstream_read_bit(bc); lossless_check = bitstream_read(bc, 8); if (substr == m->max_decoded_substream && s->lossless_check_data != 0xffffffff) { tmp = xor_32_to_8(s->lossless_check_data); if (tmp != lossless_check) av_log(m->avctx, AV_LOG_WARNING, "Lossless check failed - expected %02x, calculated %02x.\n", lossless_check, tmp); } bitstream_skip(bc, 16); memset(s->ch_assign, 0, sizeof(s->ch_assign)); for (ch = 0; ch <= s->max_matrix_channel; ch++) { int ch_assign = bitstream_read(bc, 6); if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) { uint64_t channel = thd_channel_layout_extract_channel(s->mask, ch_assign); ch_assign = av_get_channel_layout_channel_index(s->mask, channel); } if (ch_assign < 0 || ch_assign > s->max_matrix_channel) { avpriv_request_sample(m->avctx, "Assignment of matrix channel %d to invalid output channel %d", ch, ch_assign); return AVERROR_PATCHWELCOME; } s->ch_assign[ch_assign] = ch; } checksum = ff_mlp_restart_checksum(buf, bitstream_tell(bc) - start_count); if (checksum != bitstream_read(bc, 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++) { ChannelParams *cp = &s->channel_params[ch]; cp->filter_params[FIR].order = 0; cp->filter_params[IIR].order = 0; cp->filter_params[FIR].shift = 0; cp->filter_params[IIR].shift = 0; /* Default audio coding is 24-bit raw PCM. */ cp->huff_offset = 0; cp->sign_huff_offset = -(1 << 23); cp->codebook = 0; cp->huff_lsbs = 24; } if (substr == m->max_decoded_substream) { m->avctx->channels = s->max_matrix_channel + 1; m->avctx->channel_layout = s->mask; m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(s->ch_assign, s->output_shift, s->max_matrix_channel, m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); } return 0; } /** Read parameters for one of the prediction filters. */ static int read_filter_params(MLPDecodeContext *m, BitstreamContext *bc, unsigned int substr, unsigned int channel, unsigned int filter) { SubStream *s = &m->substream[substr]; FilterParams *fp = &s->channel_params[channel].filter_params[filter]; const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER; const char fchar = filter ? 'I' : 'F'; int i, order; // Filter is 0 for FIR, 1 for IIR. assert(filter < 2); if (m->filter_changed[channel][filter]++ > 1) { av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n"); return AVERROR_INVALIDDATA; } order = bitstream_read(bc, 4); if (order > max_order) { av_log(m->avctx, AV_LOG_ERROR, "%cIR filter order %d is greater than maximum %d.\n", fchar, order, max_order); return AVERROR_INVALIDDATA; } fp->order = order; if (order > 0) { int32_t *fcoeff = s->channel_params[channel].coeff[filter]; int coeff_bits, coeff_shift; fp->shift = bitstream_read(bc, 4); coeff_bits = bitstream_read(bc, 5); coeff_shift = bitstream_read(bc, 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 AVERROR_INVALIDDATA; } 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 AVERROR_INVALIDDATA; } for (i = 0; i < order; i++) fcoeff[i] = bitstream_read_signed(bc, coeff_bits) << coeff_shift; if (bitstream_read_bit(bc)) { int state_bits, state_shift; if (filter == FIR) { av_log(m->avctx, AV_LOG_ERROR, "FIR filter has state data specified.\n"); return AVERROR_INVALIDDATA; } state_bits = bitstream_read(bc, 4); state_shift = bitstream_read(bc, 4); /* TODO: Check validity of state data. */ for (i = 0; i < order; i++) fp->state[i] = bitstream_read_signed(bc, state_bits) << state_shift; } } return 0; } /** Read parameters for primitive matrices. */ static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, BitstreamContext *bc) { SubStream *s = &m->substream[substr]; unsigned int mat, ch; const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP ? MAX_MATRICES_MLP : MAX_MATRICES_TRUEHD; if (m->matrix_changed++ > 1) { av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n"); return AVERROR_INVALIDDATA; } s->num_primitive_matrices = bitstream_read(bc, 4); if (s->num_primitive_matrices > max_primitive_matrices) { av_log(m->avctx, AV_LOG_ERROR, "Number of primitive matrices cannot be greater than %d.\n", max_primitive_matrices); return AVERROR_INVALIDDATA; } for (mat = 0; mat < s->num_primitive_matrices; mat++) { int frac_bits, max_chan; s->matrix_out_ch[mat] = bitstream_read(bc, 4); frac_bits = bitstream_read(bc, 4); s->lsb_bypass[mat] = bitstream_read_bit(bc); if (s->matrix_out_ch[mat] > s->max_matrix_channel) { av_log(m->avctx, AV_LOG_ERROR, "Invalid channel %d specified as output from matrix.\n", s->matrix_out_ch[mat]); return AVERROR_INVALIDDATA; } if (frac_bits > 14) { av_log(m->avctx, AV_LOG_ERROR, "Too many fractional bits specified.\n"); return AVERROR_INVALIDDATA; } 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 (bitstream_read_bit(bc)) coeff_val = bitstream_read_signed(bc, frac_bits + 2); s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); } if (s->noise_type) s->matrix_noise_shift[mat] = bitstream_read(bc, 4); else s->matrix_noise_shift[mat] = 0; } return 0; } /** Read channel parameters. */ static int read_channel_params(MLPDecodeContext *m, unsigned int substr, BitstreamContext *bc, unsigned int ch) { SubStream *s = &m->substream[substr]; ChannelParams *cp = &s->channel_params[ch]; FilterParams *fir = &cp->filter_params[FIR]; FilterParams *iir = &cp->filter_params[IIR]; int ret; if (s->param_presence_flags & PARAM_FIR) if (bitstream_read_bit(bc)) if ((ret = read_filter_params(m, bc, substr, ch, FIR)) < 0) return ret; if (s->param_presence_flags & PARAM_IIR) if (bitstream_read_bit(bc)) if ((ret = read_filter_params(m, bc, substr, ch, IIR)) < 0) return ret; if (fir->order + iir->order > 8) { av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n"); return AVERROR_INVALIDDATA; } if (fir->order && iir->order && fir->shift != iir->shift) { av_log(m->avctx, AV_LOG_ERROR, "FIR and IIR filters must use the same precision.\n"); return AVERROR_INVALIDDATA; } /* 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 (!fir->order && iir->order) fir->shift = iir->shift; if (s->param_presence_flags & PARAM_HUFFOFFSET) if (bitstream_read_bit(bc)) cp->huff_offset = bitstream_read_signed(bc, 15); cp->codebook = bitstream_read(bc, 2); cp->huff_lsbs = bitstream_read(bc, 5); if (cp->huff_lsbs > 24) { av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n"); return AVERROR_INVALIDDATA; } cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); return 0; } /** Read decoding parameters that change more often than those in the restart * header. */ static int read_decoding_params(MLPDecodeContext *m, BitstreamContext *bc, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int ch; int ret; if (s->param_presence_flags & PARAM_PRESENCE) if (bitstream_read_bit(bc)) s->param_presence_flags = bitstream_read(bc, 8); if (s->param_presence_flags & PARAM_BLOCKSIZE) if (bitstream_read_bit(bc)) { s->blocksize = bitstream_read(bc, 9); if (s->blocksize < 8 || s->blocksize > m->access_unit_size) { av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize."); s->blocksize = 0; return AVERROR_INVALIDDATA; } } if (s->param_presence_flags & PARAM_MATRIX) if (bitstream_read_bit(bc)) if ((ret = read_matrix_params(m, substr, bc)) < 0) return ret; if (s->param_presence_flags & PARAM_OUTSHIFT) if (bitstream_read_bit(bc)) { for (ch = 0; ch <= s->max_matrix_channel; ch++) s->output_shift[ch] = bitstream_read_signed(bc, 4); if (substr == m->max_decoded_substream) m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(s->ch_assign, s->output_shift, s->max_matrix_channel, m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); } if (s->param_presence_flags & PARAM_QUANTSTEP) if (bitstream_read_bit(bc)) for (ch = 0; ch <= s->max_channel; ch++) { ChannelParams *cp = &s->channel_params[ch]; s->quant_step_size[ch] = bitstream_read(bc, 4); cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); } for (ch = s->min_channel; ch <= s->max_channel; ch++) if (bitstream_read_bit(bc)) if ((ret = read_channel_params(m, substr, bc, ch)) < 0) return ret; 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]; const int32_t *fircoeff = s->channel_params[channel].coeff[FIR]; int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER]; int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE; int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE; FilterParams *fir = &s->channel_params[channel].filter_params[FIR]; FilterParams *iir = &s->channel_params[channel].filter_params[IIR]; unsigned int filter_shift = fir->shift; int32_t mask = MSB_MASK(s->quant_step_size[channel]); memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t)); memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t)); m->dsp.mlp_filter_channel(firbuf, fircoeff, fir->order, iir->order, filter_shift, mask, s->blocksize, &m->sample_buffer[s->blockpos][channel]); memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t)); memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t)); } /** Read a block of PCM residual data (or actual if no filtering active). */ static int read_block_data(MLPDecodeContext *m, BitstreamContext *bc, unsigned int substr) { SubStream *s = &m->substream[substr]; unsigned int i, ch, expected_stream_pos = 0; int ret; if (s->data_check_present) { expected_stream_pos = bitstream_tell(bc); expected_stream_pos += bitstream_read(bc, 16); avpriv_request_sample(m->avctx, "Substreams with VLC block size check info"); } if (s->blockpos + s->blocksize > m->access_unit_size) { av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n"); return AVERROR_INVALIDDATA; } memset(&m->bypassed_lsbs[s->blockpos][0], 0, s->blocksize * sizeof(m->bypassed_lsbs[0])); for (i = 0; i < s->blocksize; i++) if ((ret = read_huff_channels(m, bc, substr, i)) < 0) return ret; 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 (bitstream_tell(bc) != expected_stream_pos) av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n"); bitstream_skip(bc, 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; 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++) { unsigned int dest_ch = s->matrix_out_ch[mat]; m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0], s->matrix_coeff[mat], &m->bypassed_lsbs[0][mat], m->noise_buffer, s->num_primitive_matrices - mat, dest_ch, s->blockpos, maxchan, s->matrix_noise_shift[mat], m->access_unit_size_pow2, MSB_MASK(s->quant_step_size[dest_ch])); } } /** Write the audio data into the output buffer. */ static int output_data(MLPDecodeContext *m, unsigned int substr, AVFrame *frame, int *got_frame_ptr) { AVCodecContext *avctx = m->avctx; SubStream *s = &m->substream[substr]; int ret; int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); if (m->avctx->channels != s->max_matrix_channel + 1) { av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n"); return AVERROR_INVALIDDATA; } if (!s->blockpos) { av_log(avctx, AV_LOG_ERROR, "No samples to output.\n"); return AVERROR_INVALIDDATA; } /* get output buffer */ frame->nb_samples = s->blockpos; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } s->lossless_check_data = m->dsp.mlp_pack_output(s->lossless_check_data, s->blockpos, m->sample_buffer, frame->data[0], s->ch_assign, s->output_shift, s->max_matrix_channel, is32); /* Update matrix encoding side data */ if ((ret = ff_side_data_update_matrix_encoding(frame, s->matrix_encoding)) < 0) return ret; *got_frame_ptr = 1; return 0; } /** Read an access unit from the stream. * @return negative on error, 0 if not enough data is present in the input stream, * otherwise the number of bytes consumed. */ static int read_access_unit(AVCodecContext *avctx, void* data, int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; MLPDecodeContext *m = avctx->priv_data; BitstreamContext bc; 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; int ret; if (buf_size < 4) return 0; length = (AV_RB16(buf) & 0xfff) * 2; if (length < 4 || length > buf_size) return AVERROR_INVALIDDATA; bitstream_init8(&bc, buf + 4, length - 4); m->is_major_sync_unit = 0; if (bitstream_peek(&bc, 31) == (0xf8726fba >> 1)) { if (read_major_sync(m, &bc) < 0) goto error; m->is_major_sync_unit = 1; header_size += m->major_sync_header_size; } if (!m->params_valid) { av_log(m->avctx, AV_LOG_WARNING, "Stream parameters not seen; skipping frame.\n"); *got_frame_ptr = 0; return length; } substream_start = 0; for (substr = 0; substr < m->num_substreams; substr++) { int extraword_present, checkdata_present, end, nonrestart_substr; extraword_present = bitstream_read_bit(&bc); nonrestart_substr = bitstream_read_bit(&bc); checkdata_present = bitstream_read_bit(&bc); bitstream_skip(&bc, 1); end = bitstream_read(&bc, 12) * 2; substr_header_size += 2; if (extraword_present) { if (m->avctx->codec_id == AV_CODEC_ID_MLP) { av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n"); goto error; } bitstream_skip(&bc, 16); substr_header_size += 2; } if (!(nonrestart_substr ^ m->is_major_sync_unit)) { av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n"); goto error; } 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 = ff_mlp_calculate_parity(buf, 4); parity_bits ^= ff_mlp_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]; bitstream_init8(&bc, buf, substream_data_len[substr]); m->matrix_changed = 0; memset(m->filter_changed, 0, sizeof(m->filter_changed)); s->blockpos = 0; do { if (bitstream_read_bit(&bc)) { if (bitstream_read_bit(&bc)) { /* A restart header should be present. */ if (read_restart_header(m, &bc, buf, substr) < 0) goto next_substr; s->restart_seen = 1; } if (!s->restart_seen) goto next_substr; if (read_decoding_params(m, &bc, substr) < 0) goto next_substr; } if (!s->restart_seen) goto next_substr; if ((ret = read_block_data(m, &bc, substr)) < 0) return ret; if (bitstream_tell(&bc) >= substream_data_len[substr] * 8) goto substream_length_mismatch; } while (!bitstream_read_bit(&bc)); bitstream_skip(&bc, (-bitstream_tell(&bc)) & 15); if (substream_data_len[substr] * 8 - bitstream_tell(&bc) >= 32) { int shorten_by; if (bitstream_read(&bc, 16) != 0xD234) return AVERROR_INVALIDDATA; shorten_by = bitstream_read(&bc, 16); if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by & 0x2000) s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos); else if (m->avctx->codec_id == AV_CODEC_ID_MLP && shorten_by != 0xD234) return AVERROR_INVALIDDATA; if (substr == m->max_decoded_substream) av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n"); } if (substream_parity_present[substr]) { uint8_t parity, checksum; if (substream_data_len[substr] * 8 - bitstream_tell(&bc) != 16) goto substream_length_mismatch; parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2); checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2); if ((bitstream_read(&bc, 8) ^ parity) != 0xa9) av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr); if (bitstream_read(&bc, 8) != checksum) av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr); } if (substream_data_len[substr] * 8 != bitstream_tell(&bc)) goto substream_length_mismatch; next_substr: if (!s->restart_seen) av_log(m->avctx, AV_LOG_ERROR, "No restart header present in substream %d.\n", substr); buf += substream_data_len[substr]; } rematrix_channels(m, m->max_decoded_substream); if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0) return ret; return length; substream_length_mismatch: av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr); return AVERROR_INVALIDDATA; error: m->params_valid = 0; return AVERROR_INVALIDDATA; } AVCodec ff_mlp_decoder = { .name = "mlp", .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MLP, .priv_data_size = sizeof(MLPDecodeContext), .init = mlp_decode_init, .decode = read_access_unit, .capabilities = AV_CODEC_CAP_DR1, }; #if CONFIG_TRUEHD_DECODER AVCodec ff_truehd_decoder = { .name = "truehd", .long_name = NULL_IF_CONFIG_SMALL("TrueHD"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_TRUEHD, .priv_data_size = sizeof(MLPDecodeContext), .init = mlp_decode_init, .decode = read_access_unit, .capabilities = AV_CODEC_CAP_DR1, }; #endif /* CONFIG_TRUEHD_DECODER */