Mercurial > libavcodec.hg
changeset 7194:8427f12555a6 libavcodec
MLP/TrueHD decoder.
| author | ramiro |
|---|---|
| date | Fri, 04 Jul 2008 15:44:13 +0000 |
| parents | 3ab9c2bc0413 |
| children | 5e5c1178ae9a |
| files | Makefile allcodecs.c avcodec.h mlpdec.c |
| diffstat | 4 files changed, 1184 insertions(+), 2 deletions(-) [+] |
line wrap: on
line diff
--- a/Makefile Fri Jul 04 15:37:52 2008 +0000 +++ b/Makefile Fri Jul 04 15:44:13 2008 +0000 @@ -107,6 +107,7 @@ OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o mjpeg.o OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o mjpegdec.o mjpeg.o +OBJS-$(CONFIG_MLP_DECODER) += mlpdec.o OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o OBJS-$(CONFIG_MP2_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o mpegaudiodata.o
--- a/allcodecs.c Fri Jul 04 15:37:52 2008 +0000 +++ b/allcodecs.c Fri Jul 04 15:44:13 2008 +0000 @@ -189,6 +189,7 @@ 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);
--- a/avcodec.h Fri Jul 04 15:37:52 2008 +0000 +++ b/avcodec.h Fri Jul 04 15:44:13 2008 +0000 @@ -30,8 +30,8 @@ #include "libavutil/avutil.h" #define LIBAVCODEC_VERSION_MAJOR 51 -#define LIBAVCODEC_VERSION_MINOR 57 -#define LIBAVCODEC_VERSION_MICRO 2 +#define LIBAVCODEC_VERSION_MINOR 58 +#define LIBAVCODEC_VERSION_MICRO 0 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ LIBAVCODEC_VERSION_MINOR, \
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/mlpdec.c Fri Jul 04 15:44:13 2008 +0000 @@ -0,0 +1,1180 @@ +/* + * 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 "libavutil/intreadwrite.h" +#include "bitstream.h" +#include "libavutil/crc.h" +#include "parser.h" +#include "mlp_parser.h" + +/** Maximum number of channels that can be decoded. */ +#define MAX_CHANNELS 16 + +/** Maximum number of matrices used in decoding. Most streams have one matrix + * per output channel, but some rematrix a channel (usually 0) more than once. + */ + +#define MAX_MATRICES 15 + +/** Maximum number of substreams that can be decoded. This could also be set + * higher, but again I haven't seen any examples with more than two. */ +#define MAX_SUBSTREAMS 2 + +/** Maximum sample frequency seen in files. */ +#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)) + +/** Number of allowed filters. */ +#define NUM_FILTERS 2 + +/** The maximum number of taps in either the IIR or FIR filter. + * I believe MLP actually specifies the maximum order for IIR filters as four, + * and that the sum of the orders of both filters must be <= 8. */ +#define MAX_FILTER_ORDER 8 + +/** Number of bits used for VLC lookup - longest huffman code is 9. */ +#define VLC_BITS 9 + + +static const char* sample_message = + "Please file a bug report following the instructions at " + "http://ffmpeg.mplayerhq.hu/bugreports.html and include " + "a sample of this file."; + +typedef struct SubStream { + //! Set if a valid restart header has been read. Otherwise the substream can not be decoded. + uint8_t restart_seen; + + //@{ + /** restart header data */ + //! The type of noise to be used in the rematrix stage. + uint16_t noise_type; + + //! The index of the first channel coded in this substream. + uint8_t min_channel; + //! The index of the last channel coded in this substream. + uint8_t max_channel; + //! The number of channels input into the rematrix stage. + uint8_t max_matrix_channel; + + //! The left shift applied to random noise in 0x31ea substreams. + uint8_t noise_shift; + //! The current seed value for the pseudorandom noise generator(s). + uint32_t noisegen_seed; + + //! Set if the substream contains extra info to check the size of VLC blocks. + uint8_t data_check_present; + + //! Bitmask of which parameter sets are conveyed in a decoding parameter block. + uint8_t param_presence_flags; +#define PARAM_BLOCKSIZE (1 << 7) +#define PARAM_MATRIX (1 << 6) +#define PARAM_OUTSHIFT (1 << 5) +#define PARAM_QUANTSTEP (1 << 4) +#define PARAM_FIR (1 << 3) +#define PARAM_IIR (1 << 2) +#define PARAM_HUFFOFFSET (1 << 1) + //@} + + //@{ + /** matrix data */ + + //! Number of matrices to be applied. + uint8_t num_primitive_matrices; + + //! matrix output channel + uint8_t matrix_out_ch[MAX_MATRICES]; + + //! Whether the LSBs of the matrix output are encoded in the bitstream. + uint8_t lsb_bypass[MAX_MATRICES]; + //! Matrix coefficients, stored as 2.14 fixed point. + int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2]; + //! Left shift to apply to noise values in 0x31eb substreams. + uint8_t matrix_noise_shift[MAX_MATRICES]; + //@} + + //! Left shift to apply to huffman-decoded residuals. + uint8_t quant_step_size[MAX_CHANNELS]; + + //! Number of PCM samples in current audio block. + uint16_t blocksize; + //! Number of PCM samples decoded so far in this frame. + uint16_t blockpos; + + //! Left shift to apply to decoded PCM values to get final 24-bit output. + int8_t output_shift[MAX_CHANNELS]; + + //! Running XOR of all output samples. + int32_t lossless_check_data; + +} SubStream; + +typedef struct MLPDecodeContext { + AVCodecContext *avctx; + + //! Set if a valid major sync block has been read. Otherwise no decoding is possible. + uint8_t params_valid; + + //! Number of substreams contained within this stream. + uint8_t num_substreams; + + //! Index of the last substream to decode - further substreams are skipped. + uint8_t max_decoded_substream; + + //! Number of PCM samples contained in each frame. + int access_unit_size; + //! Next power of two above the number of samples in each frame. + int access_unit_size_pow2; + + SubStream substream[MAX_SUBSTREAMS]; + + //@{ + /** filter data */ +#define FIR 0 +#define IIR 1 + //! Number of taps in filter. + uint8_t filter_order[MAX_CHANNELS][NUM_FILTERS]; + //! Right shift to apply to output of filter. + uint8_t filter_shift[MAX_CHANNELS][NUM_FILTERS]; + + int32_t filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER]; + int32_t filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER]; + //@} + + //@{ + /** sample data coding infomation */ + //! Offset to apply to residual values. + int16_t huff_offset[MAX_CHANNELS]; + //! Sign/rounding corrected version of huff_offset. + int32_t sign_huff_offset[MAX_CHANNELS]; + //! Which VLC codebook to use to read residuals. + uint8_t codebook[MAX_CHANNELS]; + //! Size of residual suffix not encoded using VLC. + uint8_t huff_lsbs[MAX_CHANNELS]; + //@} + + int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; + int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; + int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2]; +} MLPDecodeContext; + +/** Tables defining the huffman codes. + * There are three entropy coding methods used in MLP (four if you count + * "none" as a method). These use the same sequences for codes starting with + * 00 or 01, but have different codes starting with 1. */ + +static const uint8_t huffman_tables[3][18][2] = { + { /* huffman table 0, -7 - +10 */ + {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, + {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3}, + {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, + }, { /* huffman table 1, -7 - +8 */ + {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, + {0x02, 2}, {0x03, 2}, + {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, + }, { /* huffman table 2, -7 - +7 */ + {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, + {0x01, 1}, + {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, + } +}; + +static VLC huff_vlc[3]; + +static int crc_init = 0; +static AVCRC crc_63[1024]; +static AVCRC crc_1D[1024]; + + +/** Initialize static data, constant between all invocations of the codec. */ + +static av_cold void init_static() +{ + INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, + &huffman_tables[0][0][1], 2, 1, + &huffman_tables[0][0][0], 2, 1, 512); + INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, + &huffman_tables[1][0][1], 2, 1, + &huffman_tables[1][0][0], 2, 1, 512); + INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, + &huffman_tables[2][0][1], 2, 1, + &huffman_tables[2][0][0], 2, 1, 512); + + if (!crc_init) { + av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63)); + av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D)); + crc_init = 1; + } +} + + +/** MLP uses checksums that seem to be based on the standard CRC algorithm, + * but not (in implementation terms, the table lookup and XOR are reversed). + * We can implement this behavior using a standard av_crc on all but the + * last element, then XOR that with the last element. */ + +static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size) +{ + uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c + checksum ^= buf[buf_size-1]; + return checksum; +} + +/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8 + * number of bits, starting two bits into the first byte of buf. */ + +static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size) +{ + int i; + int num_bytes = (bit_size + 2) / 8; + + int crc = crc_1D[buf[0] & 0x3f]; + crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2); + crc ^= buf[num_bytes - 1]; + + for (i = 0; i < ((bit_size + 2) & 7); i++) { + crc <<= 1; + if (crc & 0x100) + crc ^= 0x11D; + crc ^= (buf[num_bytes] >> (7 - i)) & 1; + } + + return crc; +} + +static inline int32_t calculate_sign_huff(MLPDecodeContext *m, + unsigned int substr, unsigned int ch) +{ + SubStream *s = &m->substream[substr]; + int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch]; + int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1); + int32_t sign_huff_offset = m->huff_offset[ch]; + + if (m->codebook[ch] > 0) + sign_huff_offset -= 7 << lsb_bits; + + if (sign_shift >= 0) + sign_huff_offset -= 1 << sign_shift; + + return sign_huff_offset; +} + +/** Read a sample, consisting of either, both or neither of entropy-coded MSBs + * and plain LSBs. */ + +static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, + unsigned int substr, unsigned int pos) +{ + SubStream *s = &m->substream[substr]; + unsigned int mat, channel; + + for (mat = 0; mat < s->num_primitive_matrices; mat++) + if (s->lsb_bypass[mat]) + m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp); + + for (channel = s->min_channel; channel <= s->max_channel; channel++) { + int codebook = m->codebook[channel]; + int quant_step_size = s->quant_step_size[channel]; + int lsb_bits = m->huff_lsbs[channel] - quant_step_size; + int result = 0; + + if (codebook > 0) + result = get_vlc2(gbp, huff_vlc[codebook-1].table, + VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); + + if (result < 0) + return -1; + + if (lsb_bits > 0) + result = (result << lsb_bits) + get_bits(gbp, lsb_bits); + + result += m->sign_huff_offset[channel]; + result <<= quant_step_size; + + m->sample_buffer[pos + s->blockpos][channel] = result; + } + + return 0; +} + +static av_cold int mlp_decode_init(AVCodecContext *avctx) +{ + MLPDecodeContext *m = avctx->priv_data; + int substr; + + init_static(); + m->avctx = avctx; + for (substr = 0; substr < MAX_SUBSTREAMS; substr++) + m->substream[substr].lossless_check_data = 0xffffffff; + return 0; +} + +/** Read a major sync info header - contains high level information about + * the stream - sample rate, channel arrangement etc. Most of this + * information is not actually necessary for decoding, only for playback. + */ + +static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb) +{ + MLPHeaderInfo mh; + int substr; + + if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0) + return -1; + + if (mh.group1_bits == 0) { + av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown bits per sample\n"); + return -1; + } + if (mh.group2_bits > mh.group1_bits) { + av_log(m->avctx, AV_LOG_ERROR, + "Channel group 2 cannot have more bits per sample than group 1\n"); + return -1; + } + + if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) { + av_log(m->avctx, AV_LOG_ERROR, + "Channel groups with differing sample rates 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; + for (substr = 0; substr < MAX_SUBSTREAMS; substr++) + m->substream[substr].restart_seen = 0; + + return 0; +} + +/** Read a restart header from a block in a substream. This contains parameters + * required to decode the audio that do not change very often. Generally + * (always) present only in blocks following a major sync. */ + +static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, + const uint8_t *buf, unsigned int substr) +{ + SubStream *s = &m->substream[substr]; + unsigned int ch; + int sync_word, tmp; + uint8_t checksum; + uint8_t lossless_check; + int start_count = get_bits_count(gbp); + + sync_word = get_bits(gbp, 13); + + if (sync_word != 0x31ea >> 1) { + av_log(m->avctx, AV_LOG_ERROR, + "Restart header sync incorrect (got 0x%04x)\n", sync_word); + return -1; + } + s->noise_type = get_bits1(gbp); + + skip_bits(gbp, 16); /* Output timestamp */ + + s->min_channel = get_bits(gbp, 4); + s->max_channel = get_bits(gbp, 4); + s->max_matrix_channel = get_bits(gbp, 4); + + if (s->min_channel > s->max_channel) { + av_log(m->avctx, AV_LOG_ERROR, + "Substream min channel cannot be greater than max channel.\n"); + return -1; + } + + if (m->avctx->request_channels > 0 + && s->max_channel + 1 >= m->avctx->request_channels + && substr < m->max_decoded_substream) { + av_log(m->avctx, AV_LOG_INFO, + "Extracting %d channel downmix from substream %d. " + "Further substreams will be skipped.\n", + s->max_channel + 1, substr); + m->max_decoded_substream = substr; + } + + s->noise_shift = get_bits(gbp, 4); + s->noisegen_seed = get_bits(gbp, 23); + + skip_bits(gbp, 19); + + s->data_check_present = get_bits1(gbp); + lossless_check = get_bits(gbp, 8); + if (substr == m->max_decoded_substream + && s->lossless_check_data != 0xffffffff) { + tmp = s->lossless_check_data; + tmp ^= tmp >> 16; + tmp ^= tmp >> 8; + tmp &= 0xff; + if (tmp != lossless_check) + av_log(m->avctx, AV_LOG_WARNING, + "Lossless check failed - expected %02x, calculated %02x\n", + lossless_check, tmp); + else + dprintf(m->avctx, "Lossless check passed for substream %d (%x)\n", + substr, tmp); + } + + skip_bits(gbp, 16); + + for (ch = 0; ch <= s->max_matrix_channel; ch++) { + int ch_assign = get_bits(gbp, 6); + dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch, + ch_assign); + if (ch_assign != ch) { + av_log(m->avctx, AV_LOG_ERROR, + "Non 1:1 channel assignments are used in this stream. %s\n", + sample_message); + return -1; + } + } + + checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count); + + if (checksum != get_bits(gbp, 8)) + av_log(m->avctx, AV_LOG_ERROR, "Restart header checksum error\n"); + + /* Set default decoding parameters */ + s->param_presence_flags = 0xff; + s->num_primitive_matrices = 0; + s->blocksize = 8; + s->lossless_check_data = 0; + + memset(s->output_shift , 0, sizeof(s->output_shift )); + memset(s->quant_step_size, 0, sizeof(s->quant_step_size)); + + for (ch = s->min_channel; ch <= s->max_channel; ch++) { + m->filter_order[ch][FIR] = 0; + m->filter_order[ch][IIR] = 0; + m->filter_shift[ch][FIR] = 0; + m->filter_shift[ch][IIR] = 0; + + /* Default audio coding is 24-bit raw PCM */ + m->huff_offset [ch] = 0; + m->sign_huff_offset[ch] = (-1) << 23; + m->codebook [ch] = 0; + m->huff_lsbs [ch] = 24; + } + + if (substr == m->max_decoded_substream) { + m->avctx->channels = s->max_channel + 1; + } + + return 0; +} + +/** Read parameters for one of the prediction filters. */ + +static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, + unsigned int channel, unsigned int filter) +{ + const char fchar = filter ? 'I' : 'F'; + int i, order; + + // filter is 0 for FIR, 1 for IIR + assert(filter < 2); + + order = get_bits(gbp, 4); + if (order > MAX_FILTER_ORDER) { + av_log(m->avctx, AV_LOG_ERROR, + "%cIR filter order %d is greater than maximum %d\n", + fchar, order, MAX_FILTER_ORDER); + return -1; + } + m->filter_order[channel][filter] = order; + + if (order > 0) { + int coeff_bits, coeff_shift; + + m->filter_shift[channel][filter] = get_bits(gbp, 4); + + coeff_bits = get_bits(gbp, 5); + coeff_shift = get_bits(gbp, 3); + if (coeff_bits < 1 || coeff_bits > 16) { + av_log(m->avctx, AV_LOG_ERROR, + "%cIR filter coeff_bits must be between 1 and 16\n", + fchar); + return -1; + } + if (coeff_bits + coeff_shift > 16) { + av_log(m->avctx, AV_LOG_ERROR, + "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less\n", + fchar); + return -1; + } + + for (i = 0; i < order; i++) + m->filter_coeff[channel][filter][i] = + get_sbits(gbp, coeff_bits) << coeff_shift; + + if (get_bits1(gbp)) { + int state_bits, state_shift; + + if (filter == FIR) { + av_log(m->avctx, AV_LOG_ERROR, + "FIR filter has state data specified\n"); + return -1; + } + + state_bits = get_bits(gbp, 4); + state_shift = get_bits(gbp, 4); + + /* TODO: check validity of state data */ + + for (i = 0; i < order; i++) + m->filter_state[channel][filter][i] = + get_sbits(gbp, state_bits) << state_shift; + } + } + + return 0; +} + +/** Read decoding parameters that change more often than those in the restart + * header. */ + +static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, + unsigned int substr) +{ + SubStream *s = &m->substream[substr]; + unsigned int mat, ch; + + if (get_bits1(gbp)) + s->param_presence_flags = get_bits(gbp, 8); + + if (s->param_presence_flags & PARAM_BLOCKSIZE) + if (get_bits1(gbp)) { + s->blocksize = get_bits(gbp, 9); + if (s->blocksize > MAX_BLOCKSIZE) { + av_log(m->avctx, AV_LOG_ERROR, "Block size too large\n"); + s->blocksize = 0; + return -1; + } + } + + if (s->param_presence_flags & PARAM_MATRIX) + if (get_bits1(gbp)) { + s->num_primitive_matrices = get_bits(gbp, 4); + + for (mat = 0; mat < s->num_primitive_matrices; mat++) { + int frac_bits, max_chan; + s->matrix_out_ch[mat] = get_bits(gbp, 4); + frac_bits = get_bits(gbp, 4); + s->lsb_bypass [mat] = get_bits1(gbp); + + if (s->matrix_out_ch[mat] > s->max_channel) { + av_log(m->avctx, AV_LOG_ERROR, + "Invalid channel %d specified as output from matrix\n", + s->matrix_out_ch[mat]); + return -1; + } + if (frac_bits > 14) { + av_log(m->avctx, AV_LOG_ERROR, + "Too many fractional bits specified\n"); + return -1; + } + + max_chan = s->max_matrix_channel; + if (!s->noise_type) + max_chan+=2; + + for (ch = 0; ch <= max_chan; ch++) { + int coeff_val = 0; + if (get_bits1(gbp)) + coeff_val = get_sbits(gbp, frac_bits + 2); + + s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); + } + + if (s->noise_type) + s->matrix_noise_shift[mat] = get_bits(gbp, 4); + else + s->matrix_noise_shift[mat] = 0; + } + } + + if (s->param_presence_flags & PARAM_OUTSHIFT) + if (get_bits1(gbp)) + for (ch = 0; ch <= s->max_matrix_channel; ch++) { + s->output_shift[ch] = get_bits(gbp, 4); + dprintf(m->avctx, "output shift[%d] = %d\n", + ch, s->output_shift[ch]); + /* TODO: validate */ + } + + if (s->param_presence_flags & PARAM_QUANTSTEP) + if (get_bits1(gbp)) + for (ch = 0; ch <= s->max_channel; ch++) { + s->quant_step_size[ch] = get_bits(gbp, 4); + /* TODO: validate */ + + m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch); + } + + for (ch = s->min_channel; ch <= s->max_channel; ch++) + if (get_bits1(gbp)) { + if (s->param_presence_flags & PARAM_FIR) + if (get_bits1(gbp)) + if (read_filter_params(m, gbp, ch, FIR) < 0) + return -1; + + if (s->param_presence_flags & PARAM_IIR) + if (get_bits1(gbp)) + if (read_filter_params(m, gbp, ch, IIR) < 0) + return -1; + + if (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] && + m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) { + av_log(m->avctx, AV_LOG_ERROR, + "FIR and IIR filters must use 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) >= 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, s->blockpos); + s->blockpos -= 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); + 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"), +}; +