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eb2cd99c73
Substream parity check independently. This allows decoding of streams that have EOS but don't have Substream parity check. Originally committed as revision 14206 to svn://svn.ffmpeg.org/ffmpeg/trunk
1184 lines
39 KiB
C
1184 lines
39 KiB
C
/*
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* MLP decoder
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* Copyright (c) 2007-2008 Ian Caulfield
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file mlpdec.c
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* MLP decoder
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*/
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#include <stdint.h>
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#include "avcodec.h"
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#include "libavutil/intreadwrite.h"
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#include "bitstream.h"
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#include "libavutil/crc.h"
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#include "parser.h"
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#include "mlp_parser.h"
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/** Maximum number of channels that can be decoded. */
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#define MAX_CHANNELS 16
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/** Maximum number of matrices used in decoding; most streams have one matrix
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* per output channel, but some rematrix a channel (usually 0) more than once.
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*/
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#define MAX_MATRICES 15
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/** Maximum number of substreams that can be decoded. This could also be set
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* higher, but I haven't seen any examples with more than two. */
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#define MAX_SUBSTREAMS 2
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/** maximum sample frequency seen in files */
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#define MAX_SAMPLERATE 192000
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/** maximum number of audio samples within one access unit */
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#define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000))
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/** next power of two greater than MAX_BLOCKSIZE */
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#define MAX_BLOCKSIZE_POW2 (64 * (MAX_SAMPLERATE / 48000))
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/** number of allowed filters */
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#define NUM_FILTERS 2
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/** The maximum number of taps in either the IIR or FIR filter;
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* I believe MLP actually specifies the maximum order for IIR filters as four,
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* and that the sum of the orders of both filters must be <= 8. */
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#define MAX_FILTER_ORDER 8
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/** number of bits used for VLC lookup - longest Huffman code is 9 */
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#define VLC_BITS 9
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static const char* sample_message =
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"Please file a bug report following the instructions at "
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"http://ffmpeg.mplayerhq.hu/bugreports.html and include "
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"a sample of this file.";
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typedef struct SubStream {
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//! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
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uint8_t restart_seen;
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//@{
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/** restart header data */
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//! The type of noise to be used in the rematrix stage.
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uint16_t noise_type;
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//! The index of the first channel coded in this substream.
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uint8_t min_channel;
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//! The index of the last channel coded in this substream.
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uint8_t max_channel;
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//! The number of channels input into the rematrix stage.
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uint8_t max_matrix_channel;
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//! The left shift applied to random noise in 0x31ea substreams.
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uint8_t noise_shift;
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//! The current seed value for the pseudorandom noise generator(s).
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uint32_t noisegen_seed;
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//! Set if the substream contains extra info to check the size of VLC blocks.
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uint8_t data_check_present;
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//! Bitmask of which parameter sets are conveyed in a decoding parameter block.
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uint8_t param_presence_flags;
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#define PARAM_BLOCKSIZE (1 << 7)
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#define PARAM_MATRIX (1 << 6)
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#define PARAM_OUTSHIFT (1 << 5)
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#define PARAM_QUANTSTEP (1 << 4)
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#define PARAM_FIR (1 << 3)
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#define PARAM_IIR (1 << 2)
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#define PARAM_HUFFOFFSET (1 << 1)
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//@}
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//@{
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/** matrix data */
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//! Number of matrices to be applied.
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uint8_t num_primitive_matrices;
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//! matrix output channel
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uint8_t matrix_out_ch[MAX_MATRICES];
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//! Whether the LSBs of the matrix output are encoded in the bitstream.
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uint8_t lsb_bypass[MAX_MATRICES];
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//! Matrix coefficients, stored as 2.14 fixed point.
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int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];
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//! Left shift to apply to noise values in 0x31eb substreams.
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uint8_t matrix_noise_shift[MAX_MATRICES];
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//@}
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//! Left shift to apply to Huffman-decoded residuals.
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uint8_t quant_step_size[MAX_CHANNELS];
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//! number of PCM samples in current audio block
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uint16_t blocksize;
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//! Number of PCM samples decoded so far in this frame.
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uint16_t blockpos;
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//! Left shift to apply to decoded PCM values to get final 24-bit output.
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int8_t output_shift[MAX_CHANNELS];
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//! Running XOR of all output samples.
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int32_t lossless_check_data;
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} SubStream;
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typedef struct MLPDecodeContext {
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AVCodecContext *avctx;
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//! Set if a valid major sync block has been read. Otherwise no decoding is possible.
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uint8_t params_valid;
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//! Number of substreams contained within this stream.
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uint8_t num_substreams;
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//! Index of the last substream to decode - further substreams are skipped.
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uint8_t max_decoded_substream;
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//! number of PCM samples contained in each frame
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int access_unit_size;
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//! next power of two above the number of samples in each frame
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int access_unit_size_pow2;
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SubStream substream[MAX_SUBSTREAMS];
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//@{
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/** filter data */
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#define FIR 0
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#define IIR 1
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//! number of taps in filter
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uint8_t filter_order[MAX_CHANNELS][NUM_FILTERS];
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//! Right shift to apply to output of filter.
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uint8_t filter_shift[MAX_CHANNELS][NUM_FILTERS];
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int32_t filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
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int32_t filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
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//@}
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//@{
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/** sample data coding information */
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//! Offset to apply to residual values.
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int16_t huff_offset[MAX_CHANNELS];
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//! sign/rounding-corrected version of huff_offset
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int32_t sign_huff_offset[MAX_CHANNELS];
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//! Which VLC codebook to use to read residuals.
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uint8_t codebook[MAX_CHANNELS];
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//! Size of residual suffix not encoded using VLC.
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uint8_t huff_lsbs[MAX_CHANNELS];
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//@}
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int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
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int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
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int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
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} MLPDecodeContext;
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/** Tables defining the Huffman codes.
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* There are three entropy coding methods used in MLP (four if you count
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* "none" as a method). These use the same sequences for codes starting with
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* 00 or 01, but have different codes starting with 1. */
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static const uint8_t huffman_tables[3][18][2] = {
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{ /* Huffman table 0, -7 - +10 */
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{0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
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{0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},
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{0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
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}, { /* Huffman table 1, -7 - +8 */
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{0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
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{0x02, 2}, {0x03, 2},
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{0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
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}, { /* Huffman table 2, -7 - +7 */
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{0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
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{0x01, 1},
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{0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
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}
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};
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static VLC huff_vlc[3];
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static int crc_init = 0;
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static AVCRC crc_63[1024];
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static AVCRC crc_1D[1024];
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/** Initialize static data, constant between all invocations of the codec. */
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static av_cold void init_static()
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{
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INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
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&huffman_tables[0][0][1], 2, 1,
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&huffman_tables[0][0][0], 2, 1, 512);
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INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
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&huffman_tables[1][0][1], 2, 1,
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&huffman_tables[1][0][0], 2, 1, 512);
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INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
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&huffman_tables[2][0][1], 2, 1,
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&huffman_tables[2][0][0], 2, 1, 512);
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if (!crc_init) {
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av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63));
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av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D));
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crc_init = 1;
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}
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}
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/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
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* are not (in implementation terms, the table lookup and XOR are reversed).
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* We can implement this behavior using a standard av_crc on all but the
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* last element, then XOR that with the last element. */
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static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
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{
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uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c
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checksum ^= buf[buf_size-1];
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return checksum;
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}
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/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8
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* number of bits, starting two bits into the first byte of buf. */
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static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
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{
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int i;
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int num_bytes = (bit_size + 2) / 8;
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int crc = crc_1D[buf[0] & 0x3f];
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crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2);
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crc ^= buf[num_bytes - 1];
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for (i = 0; i < ((bit_size + 2) & 7); i++) {
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crc <<= 1;
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if (crc & 0x100)
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crc ^= 0x11D;
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crc ^= (buf[num_bytes] >> (7 - i)) & 1;
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}
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return crc;
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}
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static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
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unsigned int substr, unsigned int ch)
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{
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SubStream *s = &m->substream[substr];
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int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch];
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int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1);
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int32_t sign_huff_offset = m->huff_offset[ch];
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if (m->codebook[ch] > 0)
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sign_huff_offset -= 7 << lsb_bits;
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if (sign_shift >= 0)
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sign_huff_offset -= 1 << sign_shift;
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return sign_huff_offset;
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}
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/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
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* and plain LSBs. */
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static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
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unsigned int substr, unsigned int pos)
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{
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SubStream *s = &m->substream[substr];
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unsigned int mat, channel;
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for (mat = 0; mat < s->num_primitive_matrices; mat++)
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if (s->lsb_bypass[mat])
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m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
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for (channel = s->min_channel; channel <= s->max_channel; channel++) {
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int codebook = m->codebook[channel];
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int quant_step_size = s->quant_step_size[channel];
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int lsb_bits = m->huff_lsbs[channel] - quant_step_size;
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int result = 0;
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if (codebook > 0)
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result = get_vlc2(gbp, huff_vlc[codebook-1].table,
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VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
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if (result < 0)
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return -1;
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if (lsb_bits > 0)
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result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
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result += m->sign_huff_offset[channel];
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result <<= quant_step_size;
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m->sample_buffer[pos + s->blockpos][channel] = result;
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}
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return 0;
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}
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static av_cold int mlp_decode_init(AVCodecContext *avctx)
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{
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MLPDecodeContext *m = avctx->priv_data;
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int substr;
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init_static();
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m->avctx = avctx;
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for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
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m->substream[substr].lossless_check_data = 0xffffffff;
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return 0;
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}
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/** Read a major sync info header - contains high level information about
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* the stream - sample rate, channel arrangement etc. Most of this
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* information is not actually necessary for decoding, only for playback.
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*/
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static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
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{
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MLPHeaderInfo mh;
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int substr;
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if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
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return -1;
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if (mh.group1_bits == 0) {
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av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
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return -1;
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}
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if (mh.group2_bits > mh.group1_bits) {
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av_log(m->avctx, AV_LOG_ERROR,
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"Channel group 2 cannot have more bits per sample than group 1.\n");
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return -1;
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}
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if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
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av_log(m->avctx, AV_LOG_ERROR,
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"Channel groups with differing sample rates are not currently supported.\n");
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return -1;
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}
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if (mh.group1_samplerate == 0) {
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av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
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return -1;
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}
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if (mh.group1_samplerate > MAX_SAMPLERATE) {
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av_log(m->avctx, AV_LOG_ERROR,
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"Sampling rate %d is greater than the supported maximum (%d).\n",
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mh.group1_samplerate, MAX_SAMPLERATE);
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return -1;
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}
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if (mh.access_unit_size > MAX_BLOCKSIZE) {
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av_log(m->avctx, AV_LOG_ERROR,
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"Block size %d is greater than the supported maximum (%d).\n",
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mh.access_unit_size, MAX_BLOCKSIZE);
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return -1;
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}
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if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
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av_log(m->avctx, AV_LOG_ERROR,
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"Block size pow2 %d is greater than the supported maximum (%d).\n",
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mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
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return -1;
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}
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if (mh.num_substreams == 0)
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return -1;
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if (mh.num_substreams > MAX_SUBSTREAMS) {
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av_log(m->avctx, AV_LOG_ERROR,
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"Number of substreams %d is larger than the maximum supported "
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"by the decoder. %s\n", mh.num_substreams, sample_message);
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return -1;
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}
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m->access_unit_size = mh.access_unit_size;
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m->access_unit_size_pow2 = mh.access_unit_size_pow2;
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m->num_substreams = mh.num_substreams;
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m->max_decoded_substream = m->num_substreams - 1;
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m->avctx->sample_rate = mh.group1_samplerate;
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m->avctx->frame_size = mh.access_unit_size;
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#ifdef CONFIG_AUDIO_NONSHORT
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m->avctx->bits_per_sample = mh.group1_bits;
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if (mh.group1_bits > 16) {
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m->avctx->sample_fmt = SAMPLE_FMT_S32;
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}
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#endif
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m->params_valid = 1;
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for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
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m->substream[substr].restart_seen = 0;
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return 0;
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}
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/** Read a restart header from a block in a substream. This contains parameters
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* required to decode the audio that do not change very often. Generally
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* (always) present only in blocks following a major sync. */
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static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
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const uint8_t *buf, unsigned int substr)
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{
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SubStream *s = &m->substream[substr];
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unsigned int ch;
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int sync_word, tmp;
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uint8_t checksum;
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uint8_t lossless_check;
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int start_count = get_bits_count(gbp);
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sync_word = get_bits(gbp, 13);
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if (sync_word != 0x31ea >> 1) {
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av_log(m->avctx, AV_LOG_ERROR,
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"restart header sync incorrect (got 0x%04x)\n", sync_word);
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return -1;
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}
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s->noise_type = get_bits1(gbp);
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skip_bits(gbp, 16); /* Output timestamp */
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s->min_channel = get_bits(gbp, 4);
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s->max_channel = get_bits(gbp, 4);
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s->max_matrix_channel = get_bits(gbp, 4);
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if (s->min_channel > s->max_channel) {
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av_log(m->avctx, AV_LOG_ERROR,
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"Substream min channel cannot be greater than max channel.\n");
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return -1;
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}
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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"),
|
|
};
|
|
|