Index: /Users/andy/Code/C/ffmpeg/libavcodec/allcodecs.c===================================================================--- /Users/andy/Code/C/ffmpeg/libavcodec/allcodecs.c (revision 12450)+++ /Users/andy/Code/C/ffmpeg/libavcodec/allcodecs.c (working copy)@@ -184,6 +184,7 @@ REGISTER_DECODER (IMC, imc); REGISTER_DECODER (MACE3, mace3); REGISTER_DECODER (MACE6, mace6);+ REGISTER_DECODER (MLP, mlp); REGISTER_ENCDEC (MP2, mp2); REGISTER_DECODER (MP3, mp3); REGISTER_DECODER (MP3ADU, mp3adu);Index: /Users/andy/Code/C/ffmpeg/libavcodec/mlpdec.c===================================================================--- /Users/andy/Code/C/ffmpeg/libavcodec/mlpdec.c (revision 0)+++ /Users/andy/Code/C/ffmpeg/libavcodec/mlpdec.c (revision 0)@@ -0,0 +1,1073 @@+/*+ * MLP decoder+ * Copyright (c) 2007-2008 Ian Caulfield+ *+ * This file is part of FFmpeg.+ *+ * FFmpeg is free software; you can redistribute it and/or+ * modify it under the terms of the GNU Lesser General Public+ * License as published by the Free Software Foundation; either+ * version 2.1 of the License, or (at your option) any later version.+ *+ * FFmpeg is distributed in the hope that it will be useful,+ * but WITHOUT ANY WARRANTY; without even the implied warranty of+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU+ * Lesser General Public License for more details.+ *+ * You should have received a copy of the GNU Lesser General Public+ * License along with FFmpeg; if not, write to the Free Software+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA+ */++/**+ * @file mlpdec.c+ * MLP decoder+ */++#include "avcodec.h"+#include "intreadwrite.h"+#include "bitstream.h"+#include "crc.h"+#include "parser.h"+#include "mlp_parser.h"++/** Maximum number of channels that can be decoded+ */+#define MAX_CHANNELS 16++/** Maximum number of matrices used in decoding. Most streams have one matrix+ * per output channel, but some rematrix a channel (usually 0) more than once.+ */+#define MAX_MATRICES 15++/** Maximum number of substreams that can be decoded. This could also be set+ * higher, but again I haven't seen any examples with more than two.+ */+#define MAX_SUBSTREAMS 2++/** Maximum sample frequency supported+ */+#define MAX_SAMPLERATE 192000++/** The maximum number of audio samples within one access unit.+ */+#define MAX_BLOCKSIZE (40*( MAX_SAMPLERATE/48000))+/** the next power of two greater than MAX_BLOCKSIZE+ */+#define MAX_BLOCKSIZE_POW2 (64*(MAX_SAMPLERATE / 48000))++/** The maximum number of taps in either the IIR or FIR filter.+ * I believe MLP actually specifies the maximum order for IIR filters is four,+ * and that the sum of the orders of both filters must be <= 8. + */+#define MAX_FILTER_ORDER 8++/** Number of bits used for VLC lookup - longest huffman code is 9. + */+#define VLC_BITS 9+++static const char* sample_message =+ "Please file a bug report following the instructions at "+ "http://ffmpeg.mplayerhq.hu/bugreports.html and include "+ "a sample of this file.";+++typedef struct MLPDecodeContext {+ AVCodecContext *avctx; ///< Parent context++ uint8_t params_valid; ///< Do we have valid stream data read from a major sync block?+ uint8_t num_substreams; ///< Number of substreams contained within this stream+ uint8_t max_decoded substream; ///< Index of the last substream to decode - further substreams are skipped++ int access_unit_size; ///< Number of PCM samples contained in each frame+ int access_unit_size_pow2; ///< Next power of two above the number of samples in each frame++ uint8_t restart_seen[MAX_SUBSTREAMS]; ///< For each substream, whether a restart header has been read++ /* Restart header data */+ uint16_t restart_sync_word[MAX_SUBSTREAMS]; ///< The sync word used at the start of the last restart header+ uint8_t min_channel[MAX_SUBSTREAMS]; ///< The index of the first channel coded in this substream+ uint8_t max_channel[MAX_SUBSTREAMS]; ///< The index of the last channel coded in this substream+ uint8_t max_matrix_channel[MAX_SUBSTREAMS]; ///< The number of channels input into the rematrix stage++ uint8_t noise_shift[MAX_SUBSTREAMS]; ///< The left shift applied to random noise in 0x31ea substreams+ uint32_t noisegen_seed[MAX_SUBSTREAMS]; ///< The current seed value of the pseudorandom noise generator(s)++ uint8_t data_check_present[MAX_SUBSTREAMS]; ///< Does this substream contain extra info to check the size of VLC blocks?++ uint8_t param_presence_flags[MAX_SUBSTREAMS]; ///< Bitmask of which parameter sets are conveyed in a decoding parameter block+ /* End restart data */++ /* Matrix data */+ uint8_t num_primitive_matrices[MAX_SUBSTREAMS]; ///< Number of matrices to be applied+ uint8_t matric_ch[MAX_SUBSTREAMS][MAX_MATRICES]; ///< Output channel of matrix++ uint8_t lsb_bypass[MAX_SUBSTREAMS][MAX_MATRICES]; ///< Whether the LSBs of the matrix output are encoded in the bitstream+ int32_t matrix_coeff[MAX_SUBSTREAMS][MAX_MATRICES][MAX_CHANNELS+2]; ///< Matrix coefficients,stored as 2.14 fixed point+ uint8t_t matrix_noise_shift[MAX_SUBSTREAMS][MAX_MATRICES]; ///< Left shift to apply to noise values in 0x31eb substreams+ /* End matrix data */++ uint8_t quant_step_size[MAX_SUBSTREAMS][MAX_CHANNELS]; ///< Left shift to apply to huffman-decoded residuals+ uint16_t blocksize[MAX_SUBSTREAMS]; ///< Number of PCM samples in current audio block+ uint16_t blockpos[MAX_SUBSTREAMS]; ///< Number of PCM samples decoded so far in this frame++ int8_t output_shift[MAX_SUBSTREAMS][MAX_CHANNELS]; ///< Left shift to apply to decoded PCM values to get final 24-bit output++ /* Filter data. Filter 0 is an FIR filter, filter 1 IIR. */+ uint8_t filter_order[MAX_CHANNELS][2]; ///< Number of taps in filter+ uint8_t filter_coeff_q[MAX_CHANNELS][2]; ///< Right shift to apply to output of filter++ int32_t filter_coeff[MAX_CHANNELS][2][MAX_FILTER_ORDER]; // TODO: Document variable+ int32_t filter_state[MAX_CHANNELS][2][MAX_FILTER_ORDER]; // TODO: Document variable++ /* Sample data coding information */+ int16_t huff_offset[MAX_CHANNELS]; ///< Offest to apply to residual values+ int32_t sign_huff_offsetp[MAX_CHANNELS]; ///< Sign corrected version of huff_offset+ uint8_t codebook[MAX_CHANNELS]; ///< Which VLC codebook to use to read residuals+ uint8_t huff_lsbs[MAX_CHANNELS]; ///< Size of residual suffix not encoded using VLC+ /* End sample coding info */++ int32_t lossless_check_data[MAX_SUBSTREAMS]; ///< Running XOR of all output samples+ int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; // TODO: Document variable+ int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; // TODO: Document variable+ int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2]; // TODO: Document variable+};++/** Tables defining the huffman codes.+ * There are three entropy encoding methods used in MLP (four if you count "none"+ * as a method). These use the same sequences for codes starting 00... or 01...+ * but have different codes starting 1...+ */+static const uint8_t huffman_tables[3][18][2] = {+ { /* Huffman table 0, -7 - +10 */+ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},+ {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},+ {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},+ }, { /* Huffman table 1, -7 - +8 */+ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},+ {0x02, 2}, {0x03, 2},+ {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},+ }, { /* Huffman table 2, -7 - +7 */+ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},+ {0x01, 1},+ {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},+ }+};++static VLC huff_vlc[3]; // TODO: Document variable++static AVCRC crc_63[1024]; ///< The CRC table for 0x63+static AVCRC crc_1D[1024]; ///< The CRC table for 0x63++/** Initialize static data, constant between all invocations of the codec.+ */+static void init_static()+{+ if( !huff_vlc[0].bits ) {+ init_vlc(&huff_vlc[0], VLC_BITS,18,+ &huffman_tables[0][0][1],2,1+ &huffman_tables[0][0][0],2,1,1);+ init_vlc(&huff_vlc[1], VLC_BITS, 16,+ &huffman_tables[1][0][1],2,1,+ &huffman_tables[1][0][0],2,1,1);+ init_vlc(&huff_vlc[2],VLC_BITS,15,+ &huffman_tables[2][0][1],2,1,+ &huffman_tables[2][0][0],2,1,1);++ av_crc_init(crc_63,0,8,0x63,sizeof(crc_63));+ av_crc_init(crc_1D,0,8,0x1D,sizeof(crc_1D)); + }+}++/** MLP uses checksums that seem to be based on the standard CRC algorithm+ * but not (in implementation terms, the table lookup and XOR are reversed).+ * We can implement this behaviour using a standard av_crc on all but the+ * last element, then XOR that with the last element.+ * @param buf the data to checksum+ * @param buf_size the size of the data to checksum+ */+static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)+{+ uint8_t checksum = av_crc(crc_63,0x3c,buf,buf_size-1); //crc_63[0xa2]==0x3c+ checksum ^= buf[buf_size-1];+ return checksum;+}++/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8+ * number of bits, starting two bits into the first byte of buf.+ */+static uint8_t mlp_restart_checksum(const uint8_t *buf,unsigned int bit_size)+{+ int i;+ int num_bytes = (bit_size+2)/8;++ int crc = crc_1D[buf[0] & 0x3f];+ crc = av_crc(crc_1D,crc,buf+1,num_bytes -2);+ crc ^= buf[num_bytes-1];++ int limit = ((bit_size+2) & 7);+ for(i=0;i<limit; i++)+ {+ crc <<=1;+ if(crc & 0x100)+ crc ^= 0x11D;+ crc ^= (buf[num_bytes] >> (7-i)) & 1;+ }++ return crc;+}++/**+ * Calculate the sign-corrected version of the offset applied to residual+ * values in huffman coding.+ */+static inline void calculate_sign_huff(MLPDecodeContext *m,unsigned int substr, unsigned int ch)+{+ int lsb_bits = m->huff_lsbs[ch] - m->quant_step_size[substr][ch];+ int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch]:-1);++ m->sign_huff_offset[ch] = m->huff_offset[ch];++ if(m->codebook[ch] > 0)+ m->sign_huff_offset[ch] -= 7 << lsb_bits;++ if(sign_shift >= 0)+ m->sign_huff_offset[ch] -= 1 << sign_shift;+}+++/** Read a sample, consisting of either, both or neither of entropy-coded MSBs+ * and plain LSBs.+ */++static inline int read_huff(MLPDecodeContext *m, GetBitContext *gbp,+ unsigned int substr, unsigned int channel)+{+ int codebook = m->codebook[channel];+ int quant_step_size = m->quant_step_size[substr][channel];+ int lsb_bits = m->huff_lsbs[channel] - quant_step_size;+ int result = 0;++ if (codebook > 0)+ result = get_vlc2(gbp, huff_vlc[codebook-1].table,+ VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);++ if (lsb_bits > 0)+ result = (result << lsb_bits) + get_bits(gbp, lsb_bits);++ result += m->sign_huff_offset[channel];++ return result << quant_step_size;+}++/** Initialize the decoder. */++static int mlp_decode_init(AVCodecContext *avctx)+{+ MLPDecodeContext *m = avctx->priv_data;++ init_static();+ m->avctx = avctx;+ memset(m->lossless_check_data, 0xff, sizeof(m->lossless_check_data));+ 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, const uint8_t *buf,+ unsigned int buf_size)+{+ MLPHeaderInfo mh;++ if (ff_mlp_read_major_sync(m->avctx, &mh, buf, buf_size) != 0)+ return -1;++ if (mh.group1_bits == 0) {+ av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown bits per sample\n");+ return -1;+ }+ if (mh.group2_bits > mh.group1_bits) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Channel group 2 cannot have more bits per sample than group 1\n");+ return -1;+ }++ if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Channel groups with differing sample rates not currently supported\n");+ return -1;+ }++ if (mh.group1_samplerate == 0) {+ av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown sampling rate\n");+ return -1;+ }+ if (mh.group1_samplerate > MAX_SAMPLERATE) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Sampling rate %d is greater than maximum supported (%d)\n",+ mh.group1_samplerate, MAX_SAMPLERATE);+ return -1;+ }+ if (mh.access_unit_size > MAX_BLOCKSIZE) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Block size %d is greater than maximum supported (%d)\n",+ mh.access_unit_size, MAX_BLOCKSIZE);+ return -1;+ }+ if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Block size pow2 %d is greater than maximum supported (%d)\n",+ mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);+ return -1;+ }++ if (mh.num_substreams == 0)+ return -1;+ if (mh.num_substreams > MAX_SUBSTREAMS) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Number of substreams %d is more than maximum supported by "+ "decoder. %s\n", mh.num_substreams, sample_message);+ return -1;+ }++ m->access_unit_size = mh.access_unit_size;+ m->access_unit_size_pow2 = mh.access_unit_size_pow2;++ m->num_substreams = mh.num_substreams;+ m->max_decoded_substream = m->num_substreams - 1;++ m->avctx->sample_rate = mh.group1_samplerate;+ m->avctx->frame_size = mh.access_unit_size;++#ifdef CONFIG_AUDIO_NONSHORT+ m->avctx->bits_per_sample = mh.group1_bits;+ if (mh.group1_bits > 16) {+ m->avctx->sample_fmt = SAMPLE_FMT_S32;+ }+#endif++ m->params_valid = 1;+ memset(m->restart_seen, 0, sizeof(m->restart_seen));++ return 0;+}++/** Read a restart header from a block in a substream. This contains parameters+ * required to decode the audio that do not change very often. Generally+ * (always) present only in blocks following a major sync.+ */++static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,+ const uint8_t *buf, unsigned int substr)+{+ unsigned int ch;+ int sync_word, tmp;+ uint8_t checksum;+ uint8_t lossless_check;+ int start_count = get_bits_count(gbp);++ sync_word = get_bits(gbp, 14);++ if ((sync_word & 0x3ffe) != 0x31ea) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Restart header sync incorrect (got 0x%04x)\n", sync_word);+ return -1;+ }+ m->restart_sync_word[substr] = sync_word;++ skip_bits(gbp, 16); /* Output timestamp */++ m->min_channel[substr] = get_bits(gbp, 4);+ m->max_channel[substr] = get_bits(gbp, 4);+ m->max_matrix_channel[substr] = get_bits(gbp, 4);++ if (m->min_channel[substr] > m->max_channel[substr]) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Substream min channel cannot be greater than max channel.\n");+ m->min_channel[substr] = m->max_channel[substr]+ = m->max_matrix_channel[substr] = 0;+ return -1;+ }++ if (m->avctx->request_channels > 0+ && m->max_channel[substr] + 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",+ m->max_channel[substr] + 1, substr);+ m->max_decoded_substream = substr;+ }++ m->noise_shift[substr] = get_bits(gbp, 4);+ m->noisegen_seed[substr] = get_bits(gbp, 23);++ skip_bits(gbp, 19);++ m->data_check_present[substr] = get_bits1(gbp);+ lossless_check = get_bits(gbp, 8);+ if (substr == m->max_decoded_substream+ && m->lossless_check_data[substr] != 0xffffffff) {+ tmp = m->lossless_check_data[substr];+ tmp ^= tmp >> 16;+ tmp ^= tmp >> 8;+ tmp &= 0xff;+ if (tmp != lossless_check)+ av_log(m->avctx, AV_LOG_WARNING,+ "Lossless check failed - expected %x, calculated %x\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 <= m->max_matrix_channel[substr]; 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 */+ m->param_presence_flags[substr] = 0xff;+ m->num_primitive_matrices[substr] = 0;+ m->blocksize[substr] = 8;+ m->lossless_check_data[substr] = 0;++ memset(m->output_shift[substr], 0, sizeof(m->output_shift[substr]));+ memset(m->quant_step_size[substr], 0, sizeof(m->quant_step_size[substr]));++ for (ch = m->min_channel[substr]; ch <= m->max_channel[substr]; ch++) {+ m->filter_order [ch][0] = 0;+ m->filter_order [ch][1] = 0;+ m->filter_coeff_q[ch][0] = 0;+ m->filter_coeff_q[ch][1] = 0;++ memset(m->filter_coeff[ch], 0, sizeof(m->filter_coeff[ch]));+ memset(m->filter_state[ch], 0, sizeof(m->filter_state[ch]));++ /* 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 = m->max_channel[substr] + 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)+{+ 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",+ filter ? 'I' : 'F', order, MAX_FILTER_ORDER);+ return -1;+ }+ m->filter_order[channel][filter] = order;++ if (order > 0) {+ int coeff_bits, coeff_shift;++ m->filter_coeff_q[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",+ filter ? 'I' : 'F');+ 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",+ filter ? 'I' : 'F');+ 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 == 0) {+ 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)+{+ unsigned int mat, ch;++ if (get_bits1(gbp))+ m->param_presence_flags[substr] = get_bits(gbp, 8);++ if (m->param_presence_flags[substr] & 0x80)+ if (get_bits1(gbp)) {+ m->blocksize[substr] = get_bits(gbp, 9);+ if (m->blocksize[substr] > MAX_BLOCKSIZE) {+ av_log(m->avctx, AV_LOG_ERROR, "Block size too large\n");+ m->blocksize[substr] = 0;+ return -1;+ }+ }++ if (m->param_presence_flags[substr] & 0x40)+ if (get_bits1(gbp)) {+ m->num_primitive_matrices[substr] = get_bits(gbp, 4);++ for (mat = 0; mat < m->num_primitive_matrices[substr]; mat++) {+ int frac_bits, max_chan;+ m->matrix_ch[substr][mat] = get_bits(gbp, 4);+ frac_bits = get_bits(gbp, 4);+ m->lsb_bypass[substr][mat] = get_bits1(gbp);++ if (m->matrix_ch[substr][mat] > m->max_channel[substr]) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Invalid channel %d specified as output from matrix\n",+ m->matrix_ch[substr][mat]);+ m->matrix_ch[substr][mat] = 0;+ return -1;+ }+ if (frac_bits > 14) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Too many fractional bits specified\n");+ return -1;+ }++ max_chan = m->max_matrix_channel[substr];+ if (m->restart_sync_word[substr] == 0x31ea)+ 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);++ m->matrix_coeff[substr][mat][ch] = coeff_val << (14 - frac_bits);+ }++ if (m->restart_sync_word[substr] == 0x31eb)+ m->matrix_noise_shift[substr][mat] = get_bits(gbp, 4);+ else+ m->matrix_noise_shift[substr][mat] = 0;+ }+ }++ if (m->param_presence_flags[substr] & 0x20)+ if (get_bits1(gbp)) {+ for (ch = 0; ch <= m->max_matrix_channel[substr]; ch++) {+ m->output_shift[substr][ch] = get_bits(gbp, 4);+ dprintf(m->avctx, "output shift[%d] = %d\n",+ ch, m->output_shift[substr][ch]);+ /* TODO: validate */+ }+ }++ if (m->param_presence_flags[substr] & 0x10)+ if (get_bits1(gbp))+ for (ch = 0; ch <= m->max_channel[substr]; ch++) {+ m->quant_step_size[substr][ch] = get_bits(gbp, 4);+ /* TODO: validate */++ calculate_sign_huff(m, substr, ch);+ }++ for (ch = m->min_channel[substr]; ch <= m->max_channel[substr]; ch++)+ if (get_bits1(gbp)) {+ if (m->param_presence_flags[substr] & 0x08)+ if (get_bits1(gbp))+ if (read_filter_params(m, gbp, ch, 0) < 0)+ return -1;++ if (m->param_presence_flags[substr] & 0x04)+ if (get_bits1(gbp))+ if (read_filter_params(m, gbp, ch, 1) < 0)+ return -1;++ if (m->filter_order[ch][0] > 0 && m->filter_order[ch][1] > 0+ && m->filter_coeff_q[ch][0] != m->filter_coeff_q[ch][1]) {+ av_log(m->avctx, AV_LOG_ERROR,+ "FIR and IIR filters must use same precision\n");+ return -1;+ }++ if (m->param_presence_flags[substr] & 0x02)+ 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);++ calculate_sign_huff(m, substr, ch);++ /* TODO: validate */+ }++ return 0;+}++/** Generate a PCM sample using the prediction filters and a residual value+ * read from the data stream, and update the filter state.+ */++static int filter_sample(MLPDecodeContext *m, unsigned int substr,+ unsigned int channel, int32_t residual)+{+ unsigned int i, j;+ int64_t accum = 0;+ int32_t result;++ /* TODO: Move this code to DSPContext? */++ for (j = 0; j < 2; j++)+ for (i = 0; i < m->filter_order[channel][j]; i++)+ accum += (int64_t)m->filter_state[channel][j][i] *+ m->filter_coeff[channel][j][i];++ accum = accum >> m->filter_coeff_q[channel][0];+ result = (accum + residual)+ & ~((1 << m->quant_step_size[substr][channel]) - 1);++ memmove(&m->filter_state[channel][0][1], &m->filter_state[channel][0][0],+ sizeof(m->filter_state[channel][0][0]) * (MAX_FILTER_ORDER * 2 - 1));++ m->filter_state[channel][0][0] = result;+ m->filter_state[channel][1][0] = result - accum;++ return result;+}++/** Read a block of PCM residual (or actual if no filtering active) data.+ */++static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,+ unsigned int substr)+{+ unsigned int i, mat, ch, expected_stream_pos = 0;++ if (m->data_check_present[substr])+ expected_stream_pos = get_bits_count(gbp) + get_bits(gbp, 16);+ /* UNTESTED - find an example stream */++ if (m->blockpos[substr] + m->blocksize[substr] > m->access_unit_size) {+ av_log(m->avctx, AV_LOG_ERROR, "Too many audio samples in frame\n");+ return -1;+ }++ memset(&m->bypassed_lsbs[m->blockpos[substr]][0], 0,+ m->blocksize[substr] * MAX_CHANNELS);++ for (i = 0; i < m->blocksize[substr]; i++) {+ for (mat = 0; mat < m->num_primitive_matrices[substr]; mat++)+ if (m->lsb_bypass[substr][mat])+ m->bypassed_lsbs[i + m->blockpos[substr]][mat] = get_bits1(gbp);++ for (ch = m->min_channel[substr]; ch <= m->max_channel[substr]; ch++) {+ int32_t sample = read_huff(m, gbp, substr, ch);+ int32_t filtered = filter_sample(m, substr, ch, sample);++ m->sample_buffer[i + m->blockpos[substr]][ch] = filtered;+ }+ }++ m->blockpos[substr] += m->blocksize[substr];++ if (m->data_check_present[substr]) {+ 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 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_noise_1(MLPDecodeContext *m, unsigned int substr)+{+ unsigned int i;+ uint32_t seed = m->noisegen_seed[substr];+ unsigned int maxchan = m->max_matrix_channel[substr];++ for (i = 0; i < m->blockpos[substr]; i++) {+ uint16_t seed_shr7 = seed >> 7;+ m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << m->noise_shift[substr];+ m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << m->noise_shift[substr];++ seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);+ }++ m->noisegen_seed[substr] = seed;+}++/** Generate a block of noise, used when restart_sync_word == 0x31eb. */++static void generate_noise_2(MLPDecodeContext *m, unsigned int substr)+{+ unsigned int i;+ uint32_t seed = m->noisegen_seed[substr];++ 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);+ }++ m->noisegen_seed[substr] = seed;+}+++/** Apply the channel matrices in turn to reconstruct the original audio samples.+ */++static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)+{+ unsigned int mat, dest_ch, src_ch, i;+ unsigned int maxchan;++ maxchan = m->max_matrix_channel[substr];+ if (m->restart_sync_word[substr] == 0x31ea) {+ generate_noise_1(m, substr);+ maxchan += 2;+ } else {+ generate_noise_2(m, substr);+ }++ for (mat = 0; mat < m->num_primitive_matrices[substr]; mat++) {+ dest_ch = m->matrix_ch[substr][mat];++ /* TODO: DSPContext? */++ for (i = 0; i < m->blockpos[substr]; i++) {+ int64_t accum = 0;+ for (src_ch = 0; src_ch <= maxchan; src_ch++) {+ accum += (int64_t)m->sample_buffer[i][src_ch]+ * m->matrix_coeff[substr][mat][src_ch];+ }+ if (m->matrix_noise_shift[substr][mat]) {+ uint32_t index = m->num_primitive_matrices[substr] - mat;+ index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);+ accum += m->noise_buffer[index] << (m->matrix_noise_shift[substr][mat] + 7);+ }+ m->sample_buffer[i][dest_ch] = ((accum >> 14) & ~((1 << m->quant_step_size[substr][dest_ch]) - 1))+ + 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)+{+ unsigned int i, ch = 0;+ int32_t *data_32 = (int32_t*) data;+ int16_t *data_16 = (int16_t*) data;++ if (*data_size < m->max_channel[substr] * m->blockpos[substr]+ * (is32 ? 4 : 2))+ return -1;++ for (i = 0; i < m->blockpos[substr]; i++) {+ for (ch = 0; ch <= m->max_channel[substr]; ch++) {+ int32_t sample = m->sample_buffer[i][ch] << m->output_shift[substr][ch];+ m->lossless_check_data[substr] ^= (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;+ scratch &= 0xff;++ for (; buf < buf_end; buf++)+ scratch ^= *buf;++ return scratch;+}++/**+ * Read an access unit from the stream.+ * Returns -1 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,+ 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;+ uint8_t substream_parity_present[MAX_SUBSTREAMS];+ uint16_t substream_data_len[MAX_SUBSTREAMS];++ if (buf_size < 2)+ return 0;++ length = AV_RB16(buf) & 0xfff;++ if (length * 2 > buf_size)+ return -1;++ init_get_bits(&gb, buf, length * 16);+ skip_bits_long(&gb, 32);++ if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {+ dprintf(m->avctx, "Found major sync\n");+ if (read_major_sync(m, buf + 4, buf_size - 4) < 0)+ goto error;+ skip_bits_long(&gb, 28 * 8);+ }++ if (!m->params_valid) {+ av_log(m->avctx, AV_LOG_WARNING,+ "Stream parameters not seen; skipping frame");+ return length * 2;+ }++ header_size = get_bits_count(&gb) >> 4;+ 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);++ if (extraword_present)+ skip_bits(&gb, 16);++ if (end + header_size > length) {+ av_log(m->avctx, AV_LOG_ERROR,+ "Substream %d data indicated length goes off end of packet.",+ substr);+ end = length - header_size;+ }++ if (substr > m->max_decoded_substream)+ continue;++ substream_parity_present[substr] = checkdata_present;+ substream_data_len[substr] = end - substream_start;+ substream_start = end;+ }++ buf += get_bits_count(&gb) >> 3;+ buf_size -= get_bits_count(&gb) >> 3;++ for (substr = 0; substr <= m->max_decoded_substream; substr++) {+ init_get_bits(&gb, buf, substream_data_len[substr] * 16);++ m->blockpos[substr] = 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 error;+ m->restart_seen[substr] = 1;+ }++ if (!m->restart_seen[substr]) {+ av_log(m->avctx, AV_LOG_ERROR,+ "No restart header present in substream %d.\n",+ substr);+ goto error;+ }++ if (read_decoding_params(m, &gb, substr) < 0)+ goto error;+ }++ if (!m->restart_seen[substr]) {+ av_log(m->avctx, AV_LOG_ERROR,+ "No restart header present in substream %d.\n",+ substr);+ goto error;+ }++ if (read_block_data(m, &gb, substr) < 0)+ return -1;++ } while ((get_bits_count(&gb) < substream_data_len[substr] * 16)+ && get_bits1(&gb) == 0);++ skip_bits(&gb, (-get_bits_count(&gb)) & 15);+ if ((substream_data_len[substr] * 16) - get_bits_count(&gb) >= 48 &&+ (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, m->blockpos[substr]);+ m->blockpos[substr] -= shorten_by;+ } else+ skip_bits(&gb, 13);+ }+ if (substream_parity_present[substr]) {+ uint8_t parity, checksum;++ parity = calculate_parity(buf, substream_data_len[substr] * 2 - 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 - 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] * 16 != get_bits_count(&gb)) {+ av_log(m->avctx, AV_LOG_ERROR, "Substream %d length mismatch.\n",+ substr);+ return -1;+ }++ buf += substream_data_len[substr] * 2;+ buf_size -= substream_data_len[substr] * 2;+ }++ rematrix_channels(m, substr - 1);++ if (output_data(m, substr - 1, data, data_size) < 0)+ return -1;++ return length * 2;++error:+ m->params_valid = 0;+ return -1;+}++AVCodec mlp_decoder = {+ .name = "mlp",+ .type = CODEC_TYPE_AUDIO,+ .id = CODEC_ID_MLP,+ .priv_data_size = sizeof(MLPDecodeContext),+ .init = mlp_decode_init,+ .encode = NULL,+ .close = NULL,+ .decode = read_access_unit,+};+