| /* |
| * 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 |
| * MLP decoder |
| */ |
| |
| #include <stdint.h> |
| |
| #include "avcodec.h" |
| #include "libavutil/internal.h" |
| #include "libavutil/intreadwrite.h" |
| #include "libavutil/channel_layout.h" |
| #include "get_bits.h" |
| #include "internal.h" |
| #include "libavutil/crc.h" |
| #include "parser.h" |
| #include "mlp_parser.h" |
| #include "mlpdsp.h" |
| #include "mlp.h" |
| #include "config.h" |
| |
| /** number of bits used for VLC lookup - longest Huffman code is 9 */ |
| #if ARCH_ARM |
| #define VLC_BITS 5 |
| #define VLC_STATIC_SIZE 64 |
| #else |
| #define VLC_BITS 9 |
| #define VLC_STATIC_SIZE 512 |
| #endif |
| |
| typedef struct SubStream { |
| /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded. |
| uint8_t restart_seen; |
| |
| //@{ |
| /** restart header data */ |
| /// The type of noise to be used in the rematrix stage. |
| uint16_t noise_type; |
| |
| /// The index of the first channel coded in this substream. |
| uint8_t min_channel; |
| /// The index of the last channel coded in this substream. |
| uint8_t max_channel; |
| /// The number of channels input into the rematrix stage. |
| uint8_t max_matrix_channel; |
| /// For each channel output by the matrix, the output channel to map it to |
| uint8_t ch_assign[MAX_CHANNELS]; |
| /// The channel layout for this substream |
| uint64_t ch_layout; |
| /// The matrix encoding mode for this substream |
| enum AVMatrixEncoding matrix_encoding; |
| |
| /// Channel coding parameters for channels in the substream |
| ChannelParams channel_params[MAX_CHANNELS]; |
| |
| /// The left shift applied to random noise in 0x31ea substreams. |
| uint8_t noise_shift; |
| /// The current seed value for the pseudorandom noise generator(s). |
| uint32_t noisegen_seed; |
| |
| /// Set if the substream contains extra info to check the size of VLC blocks. |
| uint8_t data_check_present; |
| |
| /// Bitmask of which parameter sets are conveyed in a decoding parameter block. |
| uint8_t param_presence_flags; |
| #define PARAM_BLOCKSIZE (1 << 7) |
| #define PARAM_MATRIX (1 << 6) |
| #define PARAM_OUTSHIFT (1 << 5) |
| #define PARAM_QUANTSTEP (1 << 4) |
| #define PARAM_FIR (1 << 3) |
| #define PARAM_IIR (1 << 2) |
| #define PARAM_HUFFOFFSET (1 << 1) |
| #define PARAM_PRESENCE (1 << 0) |
| //@} |
| |
| //@{ |
| /** matrix data */ |
| |
| /// Number of matrices to be applied. |
| uint8_t num_primitive_matrices; |
| |
| /// matrix output channel |
| uint8_t matrix_out_ch[MAX_MATRICES]; |
| |
| /// Whether the LSBs of the matrix output are encoded in the bitstream. |
| uint8_t lsb_bypass[MAX_MATRICES]; |
| /// Matrix coefficients, stored as 2.14 fixed point. |
| DECLARE_ALIGNED(32, int32_t, matrix_coeff)[MAX_MATRICES][MAX_CHANNELS]; |
| /// Left shift to apply to noise values in 0x31eb substreams. |
| uint8_t matrix_noise_shift[MAX_MATRICES]; |
| //@} |
| |
| /// Left shift to apply to Huffman-decoded residuals. |
| uint8_t quant_step_size[MAX_CHANNELS]; |
| |
| /// number of PCM samples in current audio block |
| uint16_t blocksize; |
| /// Number of PCM samples decoded so far in this frame. |
| uint16_t blockpos; |
| |
| /// Left shift to apply to decoded PCM values to get final 24-bit output. |
| int8_t output_shift[MAX_CHANNELS]; |
| |
| /// Running XOR of all output samples. |
| int32_t lossless_check_data; |
| |
| } SubStream; |
| |
| typedef struct MLPDecodeContext { |
| AVCodecContext *avctx; |
| |
| /// Current access unit being read has a major sync. |
| int is_major_sync_unit; |
| |
| /// Size of the major sync unit, in bytes |
| int major_sync_header_size; |
| |
| /// Set if a valid major sync block has been read. Otherwise no decoding is possible. |
| uint8_t params_valid; |
| |
| /// Number of substreams contained within this stream. |
| uint8_t num_substreams; |
| |
| /// Index of the last substream to decode - further substreams are skipped. |
| uint8_t max_decoded_substream; |
| |
| /// Stream needs channel reordering to comply with FFmpeg's channel order |
| uint8_t needs_reordering; |
| |
| /// number of PCM samples contained in each frame |
| int access_unit_size; |
| /// next power of two above the number of samples in each frame |
| int access_unit_size_pow2; |
| |
| SubStream substream[MAX_SUBSTREAMS]; |
| |
| int matrix_changed; |
| int filter_changed[MAX_CHANNELS][NUM_FILTERS]; |
| |
| int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; |
| int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; |
| DECLARE_ALIGNED(32, int32_t, sample_buffer)[MAX_BLOCKSIZE][MAX_CHANNELS]; |
| |
| MLPDSPContext dsp; |
| } MLPDecodeContext; |
| |
| static const uint64_t thd_channel_order[] = { |
| AV_CH_FRONT_LEFT, AV_CH_FRONT_RIGHT, // LR |
| AV_CH_FRONT_CENTER, // C |
| AV_CH_LOW_FREQUENCY, // LFE |
| AV_CH_SIDE_LEFT, AV_CH_SIDE_RIGHT, // LRs |
| AV_CH_TOP_FRONT_LEFT, AV_CH_TOP_FRONT_RIGHT, // LRvh |
| AV_CH_FRONT_LEFT_OF_CENTER, AV_CH_FRONT_RIGHT_OF_CENTER, // LRc |
| AV_CH_BACK_LEFT, AV_CH_BACK_RIGHT, // LRrs |
| AV_CH_BACK_CENTER, // Cs |
| AV_CH_TOP_CENTER, // Ts |
| AV_CH_SURROUND_DIRECT_LEFT, AV_CH_SURROUND_DIRECT_RIGHT, // LRsd |
| AV_CH_WIDE_LEFT, AV_CH_WIDE_RIGHT, // LRw |
| AV_CH_TOP_FRONT_CENTER, // Cvh |
| AV_CH_LOW_FREQUENCY_2, // LFE2 |
| }; |
| |
| static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout, |
| int index) |
| { |
| int i; |
| |
| if (av_get_channel_layout_nb_channels(channel_layout) <= index) |
| return 0; |
| |
| for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++) |
| if (channel_layout & thd_channel_order[i] && !index--) |
| return thd_channel_order[i]; |
| return 0; |
| } |
| |
| static VLC huff_vlc[3]; |
| |
| /** Initialize static data, constant between all invocations of the codec. */ |
| |
| static av_cold void init_static(void) |
| { |
| if (!huff_vlc[0].bits) { |
| INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, |
| &ff_mlp_huffman_tables[0][0][1], 2, 1, |
| &ff_mlp_huffman_tables[0][0][0], 2, 1, VLC_STATIC_SIZE); |
| INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, |
| &ff_mlp_huffman_tables[1][0][1], 2, 1, |
| &ff_mlp_huffman_tables[1][0][0], 2, 1, VLC_STATIC_SIZE); |
| INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, |
| &ff_mlp_huffman_tables[2][0][1], 2, 1, |
| &ff_mlp_huffman_tables[2][0][0], 2, 1, VLC_STATIC_SIZE); |
| } |
| |
| ff_mlp_init_crc(); |
| } |
| |
| static inline int32_t calculate_sign_huff(MLPDecodeContext *m, |
| unsigned int substr, unsigned int ch) |
| { |
| SubStream *s = &m->substream[substr]; |
| ChannelParams *cp = &s->channel_params[ch]; |
| int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch]; |
| int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1); |
| int32_t sign_huff_offset = cp->huff_offset; |
| |
| if (cp->codebook > 0) |
| sign_huff_offset -= 7 << lsb_bits; |
| |
| if (sign_shift >= 0) |
| sign_huff_offset -= 1 << sign_shift; |
| |
| return sign_huff_offset; |
| } |
| |
| /** Read a sample, consisting of either, both or neither of entropy-coded MSBs |
| * and plain LSBs. */ |
| |
| static inline int read_huff_channels(MLPDecodeContext *m, 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++) { |
| ChannelParams *cp = &s->channel_params[channel]; |
| int codebook = cp->codebook; |
| int quant_step_size = s->quant_step_size[channel]; |
| int lsb_bits = cp->huff_lsbs - quant_step_size; |
| int result = 0; |
| |
| if (codebook > 0) |
| result = get_vlc2(gbp, huff_vlc[codebook-1].table, |
| VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); |
| |
| if (result < 0) |
| return AVERROR_INVALIDDATA; |
| |
| if (lsb_bits > 0) |
| result = (result << lsb_bits) + get_bits(gbp, lsb_bits); |
| |
| result += cp->sign_huff_offset; |
| result <<= quant_step_size; |
| |
| m->sample_buffer[pos + s->blockpos][channel] = result; |
| } |
| |
| return 0; |
| } |
| |
| static av_cold int mlp_decode_init(AVCodecContext *avctx) |
| { |
| MLPDecodeContext *m = avctx->priv_data; |
| int substr; |
| |
| init_static(); |
| m->avctx = avctx; |
| for (substr = 0; substr < MAX_SUBSTREAMS; substr++) |
| m->substream[substr].lossless_check_data = 0xffffffff; |
| ff_mlpdsp_init(&m->dsp); |
| |
| return 0; |
| } |
| |
| /** Read a major sync info header - contains high level information about |
| * the stream - sample rate, channel arrangement etc. Most of this |
| * information is not actually necessary for decoding, only for playback. |
| */ |
| |
| static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb) |
| { |
| MLPHeaderInfo mh; |
| int substr, ret; |
| |
| if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0) |
| return ret; |
| |
| if (mh.group1_bits == 0) { |
| av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| if (mh.group2_bits > mh.group1_bits) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Channel group 2 cannot have more bits per sample than group 1.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Channel groups with differing sample rates are not currently supported.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (mh.group1_samplerate == 0) { |
| av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| if (mh.group1_samplerate > MAX_SAMPLERATE) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Sampling rate %d is greater than the supported maximum (%d).\n", |
| mh.group1_samplerate, MAX_SAMPLERATE); |
| return AVERROR_INVALIDDATA; |
| } |
| if (mh.access_unit_size > MAX_BLOCKSIZE) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Block size %d is greater than the supported maximum (%d).\n", |
| mh.access_unit_size, MAX_BLOCKSIZE); |
| return AVERROR_INVALIDDATA; |
| } |
| if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Block size pow2 %d is greater than the supported maximum (%d).\n", |
| mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (mh.num_substreams == 0) |
| return AVERROR_INVALIDDATA; |
| if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) { |
| av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| if (mh.num_substreams > MAX_SUBSTREAMS) { |
| avpriv_request_sample(m->avctx, |
| "%d substreams (more than the " |
| "maximum supported by the decoder)", |
| mh.num_substreams); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| m->major_sync_header_size = mh.header_size; |
| |
| m->access_unit_size = mh.access_unit_size; |
| m->access_unit_size_pow2 = mh.access_unit_size_pow2; |
| |
| m->num_substreams = mh.num_substreams; |
| |
| /* limit to decoding 3 substreams, as the 4th is used by Dolby Atmos for non-audio data */ |
| m->max_decoded_substream = FFMIN(m->num_substreams - 1, 2); |
| |
| m->avctx->sample_rate = mh.group1_samplerate; |
| m->avctx->frame_size = mh.access_unit_size; |
| |
| m->avctx->bits_per_raw_sample = mh.group1_bits; |
| if (mh.group1_bits > 16) |
| m->avctx->sample_fmt = AV_SAMPLE_FMT_S32; |
| else |
| m->avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
| m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(m->substream[m->max_decoded_substream].ch_assign, |
| m->substream[m->max_decoded_substream].output_shift, |
| m->substream[m->max_decoded_substream].max_matrix_channel, |
| m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); |
| |
| m->params_valid = 1; |
| for (substr = 0; substr < MAX_SUBSTREAMS; substr++) |
| m->substream[substr].restart_seen = 0; |
| |
| /* Set the layout for each substream. When there's more than one, the first |
| * substream is Stereo. Subsequent substreams' layouts are indicated in the |
| * major sync. */ |
| if (m->avctx->codec_id == AV_CODEC_ID_MLP) { |
| if (mh.stream_type != 0xbb) { |
| avpriv_request_sample(m->avctx, |
| "unexpected stream_type %X in MLP", |
| mh.stream_type); |
| return AVERROR_PATCHWELCOME; |
| } |
| if ((substr = (mh.num_substreams > 1))) |
| m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO; |
| m->substream[substr].ch_layout = mh.channel_layout_mlp; |
| } else { |
| if (mh.stream_type != 0xba) { |
| avpriv_request_sample(m->avctx, |
| "unexpected stream_type %X in !MLP", |
| mh.stream_type); |
| return AVERROR_PATCHWELCOME; |
| } |
| if ((substr = (mh.num_substreams > 1))) |
| m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO; |
| if (mh.num_substreams > 2) |
| if (mh.channel_layout_thd_stream2) |
| m->substream[2].ch_layout = mh.channel_layout_thd_stream2; |
| else |
| m->substream[2].ch_layout = mh.channel_layout_thd_stream1; |
| m->substream[substr].ch_layout = mh.channel_layout_thd_stream1; |
| |
| if (m->avctx->channels<=2 && m->substream[substr].ch_layout == AV_CH_LAYOUT_MONO && m->max_decoded_substream == 1) { |
| av_log(m->avctx, AV_LOG_DEBUG, "Mono stream with 2 substreams, ignoring 2nd\n"); |
| m->max_decoded_substream = 0; |
| if (m->avctx->channels==2) |
| m->avctx->channel_layout = AV_CH_LAYOUT_STEREO; |
| } |
| } |
| |
| m->needs_reordering = mh.channel_arrangement >= 18 && mh.channel_arrangement <= 20; |
| |
| /* Parse the TrueHD decoder channel modifiers and set each substream's |
| * AVMatrixEncoding accordingly. |
| * |
| * The meaning of the modifiers depends on the channel layout: |
| * |
| * - THD_CH_MODIFIER_LTRT, THD_CH_MODIFIER_LBINRBIN only apply to 2-channel |
| * |
| * - THD_CH_MODIFIER_MONO applies to 1-channel or 2-channel (dual mono) |
| * |
| * - THD_CH_MODIFIER_SURROUNDEX, THD_CH_MODIFIER_NOTSURROUNDEX only apply to |
| * layouts with an Ls/Rs channel pair |
| */ |
| for (substr = 0; substr < MAX_SUBSTREAMS; substr++) |
| m->substream[substr].matrix_encoding = AV_MATRIX_ENCODING_NONE; |
| if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) { |
| if (mh.num_substreams > 2 && |
| mh.channel_layout_thd_stream2 & AV_CH_SIDE_LEFT && |
| mh.channel_layout_thd_stream2 & AV_CH_SIDE_RIGHT && |
| mh.channel_modifier_thd_stream2 == THD_CH_MODIFIER_SURROUNDEX) |
| m->substream[2].matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX; |
| |
| if (mh.num_substreams > 1 && |
| mh.channel_layout_thd_stream1 & AV_CH_SIDE_LEFT && |
| mh.channel_layout_thd_stream1 & AV_CH_SIDE_RIGHT && |
| mh.channel_modifier_thd_stream1 == THD_CH_MODIFIER_SURROUNDEX) |
| m->substream[1].matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX; |
| |
| if (mh.num_substreams > 0) |
| switch (mh.channel_modifier_thd_stream0) { |
| case THD_CH_MODIFIER_LTRT: |
| m->substream[0].matrix_encoding = AV_MATRIX_ENCODING_DOLBY; |
| break; |
| case THD_CH_MODIFIER_LBINRBIN: |
| m->substream[0].matrix_encoding = AV_MATRIX_ENCODING_DOLBYHEADPHONE; |
| break; |
| default: |
| break; |
| } |
| } |
| |
| return 0; |
| } |
| |
| /** Read a restart header from a block in a substream. This contains parameters |
| * required to decode the audio that do not change very often. Generally |
| * (always) present only in blocks following a major sync. */ |
| |
| static int read_restart_header(MLPDecodeContext *m, 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); |
| int min_channel, max_channel, max_matrix_channel; |
| const int std_max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP |
| ? MAX_MATRIX_CHANNEL_MLP |
| : MAX_MATRIX_CHANNEL_TRUEHD; |
| |
| sync_word = 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 AVERROR_INVALIDDATA; |
| } |
| |
| s->noise_type = get_bits1(gbp); |
| |
| if (m->avctx->codec_id == AV_CODEC_ID_MLP && s->noise_type) { |
| av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| skip_bits(gbp, 16); /* Output timestamp */ |
| |
| min_channel = get_bits(gbp, 4); |
| max_channel = get_bits(gbp, 4); |
| max_matrix_channel = get_bits(gbp, 4); |
| |
| if (max_matrix_channel > std_max_matrix_channel) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Max matrix channel cannot be greater than %d.\n", |
| std_max_matrix_channel); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (max_channel != max_matrix_channel) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Max channel must be equal max matrix channel.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| /* This should happen for TrueHD streams with >6 channels and MLP's noise |
| * type. It is not yet known if this is allowed. */ |
| if (max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) { |
| avpriv_request_sample(m->avctx, |
| "%d channels (more than the " |
| "maximum supported by the decoder)", |
| max_channel + 2); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| if (min_channel > max_channel) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Substream min channel cannot be greater than max channel.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| s->min_channel = min_channel; |
| s->max_channel = max_channel; |
| s->max_matrix_channel = max_matrix_channel; |
| |
| if (m->avctx->request_channel_layout && (s->ch_layout & m->avctx->request_channel_layout) == |
| m->avctx->request_channel_layout && m->max_decoded_substream > substr) { |
| av_log(m->avctx, AV_LOG_DEBUG, |
| "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. " |
| "Further substreams will be skipped.\n", |
| s->max_channel + 1, s->ch_layout, 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 = xor_32_to_8(s->lossless_check_data); |
| if (tmp != lossless_check) |
| av_log(m->avctx, AV_LOG_WARNING, |
| "Lossless check failed - expected %02x, calculated %02x.\n", |
| lossless_check, tmp); |
| } |
| |
| skip_bits(gbp, 16); |
| |
| memset(s->ch_assign, 0, sizeof(s->ch_assign)); |
| |
| for (ch = 0; ch <= s->max_matrix_channel; ch++) { |
| int ch_assign = get_bits(gbp, 6); |
| if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) { |
| uint64_t channel = thd_channel_layout_extract_channel(s->ch_layout, |
| ch_assign); |
| ch_assign = av_get_channel_layout_channel_index(s->ch_layout, |
| channel); |
| } |
| if (ch_assign < 0 || ch_assign > s->max_matrix_channel) { |
| avpriv_request_sample(m->avctx, |
| "Assignment of matrix channel %d to invalid output channel %d", |
| ch, ch_assign); |
| return AVERROR_PATCHWELCOME; |
| } |
| s->ch_assign[ch_assign] = ch; |
| } |
| |
| checksum = ff_mlp_restart_checksum(buf, 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++) { |
| ChannelParams *cp = &s->channel_params[ch]; |
| cp->filter_params[FIR].order = 0; |
| cp->filter_params[IIR].order = 0; |
| cp->filter_params[FIR].shift = 0; |
| cp->filter_params[IIR].shift = 0; |
| |
| /* Default audio coding is 24-bit raw PCM. */ |
| cp->huff_offset = 0; |
| cp->sign_huff_offset = -(1 << 23); |
| cp->codebook = 0; |
| cp->huff_lsbs = 24; |
| } |
| |
| if (substr == m->max_decoded_substream) { |
| m->avctx->channels = s->max_matrix_channel + 1; |
| m->avctx->channel_layout = s->ch_layout; |
| m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(s->ch_assign, |
| s->output_shift, |
| s->max_matrix_channel, |
| m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); |
| |
| if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) { |
| if (m->avctx->channel_layout == (AV_CH_LAYOUT_QUAD|AV_CH_LOW_FREQUENCY) || |
| m->avctx->channel_layout == AV_CH_LAYOUT_5POINT0_BACK) { |
| int i = s->ch_assign[4]; |
| s->ch_assign[4] = s->ch_assign[3]; |
| s->ch_assign[3] = s->ch_assign[2]; |
| s->ch_assign[2] = i; |
| } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) { |
| FFSWAP(int, s->ch_assign[2], s->ch_assign[4]); |
| FFSWAP(int, s->ch_assign[3], s->ch_assign[5]); |
| } |
| } |
| |
| } |
| |
| return 0; |
| } |
| |
| /** Read parameters for one of the prediction filters. */ |
| |
| static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, |
| unsigned int substr, unsigned int channel, |
| unsigned int filter) |
| { |
| SubStream *s = &m->substream[substr]; |
| FilterParams *fp = &s->channel_params[channel].filter_params[filter]; |
| const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER; |
| const char fchar = filter ? 'I' : 'F'; |
| int i, order; |
| |
| // Filter is 0 for FIR, 1 for IIR. |
| av_assert0(filter < 2); |
| |
| if (m->filter_changed[channel][filter]++ > 1) { |
| av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| order = get_bits(gbp, 4); |
| if (order > max_order) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "%cIR filter order %d is greater than maximum %d.\n", |
| fchar, order, max_order); |
| return AVERROR_INVALIDDATA; |
| } |
| fp->order = order; |
| |
| if (order > 0) { |
| int32_t *fcoeff = s->channel_params[channel].coeff[filter]; |
| int coeff_bits, coeff_shift; |
| |
| fp->shift = 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 AVERROR_INVALIDDATA; |
| } |
| if (coeff_bits + coeff_shift > 16) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n", |
| fchar); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| for (i = 0; i < order; i++) |
| fcoeff[i] = 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 AVERROR_INVALIDDATA; |
| } |
| |
| state_bits = get_bits(gbp, 4); |
| state_shift = get_bits(gbp, 4); |
| |
| /* TODO: Check validity of state data. */ |
| |
| for (i = 0; i < order; i++) |
| fp->state[i] = state_bits ? get_sbits(gbp, state_bits) << state_shift : 0; |
| } |
| } |
| |
| return 0; |
| } |
| |
| /** Read parameters for primitive matrices. */ |
| |
| static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp) |
| { |
| SubStream *s = &m->substream[substr]; |
| unsigned int mat, ch; |
| const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP |
| ? MAX_MATRICES_MLP |
| : MAX_MATRICES_TRUEHD; |
| |
| if (m->matrix_changed++ > 1) { |
| av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| s->num_primitive_matrices = get_bits(gbp, 4); |
| |
| if (s->num_primitive_matrices > max_primitive_matrices) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Number of primitive matrices cannot be greater than %d.\n", |
| max_primitive_matrices); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| for (mat = 0; mat < s->num_primitive_matrices; mat++) { |
| int frac_bits, max_chan; |
| s->matrix_out_ch[mat] = 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_matrix_channel) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Invalid channel %d specified as output from matrix.\n", |
| s->matrix_out_ch[mat]); |
| return AVERROR_INVALIDDATA; |
| } |
| if (frac_bits > 14) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Too many fractional bits specified.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| max_chan = s->max_matrix_channel; |
| if (!s->noise_type) |
| max_chan+=2; |
| |
| for (ch = 0; ch <= max_chan; ch++) { |
| int coeff_val = 0; |
| if (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; |
| } |
| |
| return 0; |
| } |
| |
| /** Read channel parameters. */ |
| |
| static int read_channel_params(MLPDecodeContext *m, unsigned int substr, |
| GetBitContext *gbp, unsigned int ch) |
| { |
| SubStream *s = &m->substream[substr]; |
| ChannelParams *cp = &s->channel_params[ch]; |
| FilterParams *fir = &cp->filter_params[FIR]; |
| FilterParams *iir = &cp->filter_params[IIR]; |
| int ret; |
| |
| if (s->param_presence_flags & PARAM_FIR) |
| if (get_bits1(gbp)) |
| if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0) |
| return ret; |
| |
| if (s->param_presence_flags & PARAM_IIR) |
| if (get_bits1(gbp)) |
| if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0) |
| return ret; |
| |
| if (fir->order + iir->order > 8) { |
| av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (fir->order && iir->order && |
| fir->shift != iir->shift) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "FIR and IIR filters must use the same precision.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| /* The FIR and IIR filters must have the same precision. |
| * To simplify the filtering code, only the precision of the |
| * FIR filter is considered. If only the IIR filter is employed, |
| * the FIR filter precision is set to that of the IIR filter, so |
| * that the filtering code can use it. */ |
| if (!fir->order && iir->order) |
| fir->shift = iir->shift; |
| |
| if (s->param_presence_flags & PARAM_HUFFOFFSET) |
| if (get_bits1(gbp)) |
| cp->huff_offset = get_sbits(gbp, 15); |
| |
| cp->codebook = get_bits(gbp, 2); |
| cp->huff_lsbs = get_bits(gbp, 5); |
| |
| if (cp->huff_lsbs > 24) { |
| av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n"); |
| cp->huff_lsbs = 0; |
| return AVERROR_INVALIDDATA; |
| } |
| |
| cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
| |
| return 0; |
| } |
| |
| /** Read decoding parameters that change more often than those in the restart |
| * header. */ |
| |
| static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, |
| unsigned int substr) |
| { |
| SubStream *s = &m->substream[substr]; |
| unsigned int ch; |
| int ret; |
| |
| if (s->param_presence_flags & PARAM_PRESENCE) |
| 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 < 8 || s->blocksize > m->access_unit_size) { |
| av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n"); |
| s->blocksize = 0; |
| return AVERROR_INVALIDDATA; |
| } |
| } |
| |
| if (s->param_presence_flags & PARAM_MATRIX) |
| if (get_bits1(gbp)) |
| if ((ret = read_matrix_params(m, substr, gbp)) < 0) |
| return ret; |
| |
| 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_sbits(gbp, 4); |
| if (substr == m->max_decoded_substream) |
| m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(s->ch_assign, |
| s->output_shift, |
| s->max_matrix_channel, |
| m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); |
| } |
| |
| if (s->param_presence_flags & PARAM_QUANTSTEP) |
| if (get_bits1(gbp)) |
| for (ch = 0; ch <= s->max_channel; ch++) { |
| ChannelParams *cp = &s->channel_params[ch]; |
| |
| s->quant_step_size[ch] = get_bits(gbp, 4); |
| |
| cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
| } |
| |
| for (ch = s->min_channel; ch <= s->max_channel; ch++) |
| if (get_bits1(gbp)) |
| if ((ret = read_channel_params(m, substr, gbp, ch)) < 0) |
| return ret; |
| |
| return 0; |
| } |
| |
| #define MSB_MASK(bits) (-1u << (bits)) |
| |
| /** Generate PCM samples using the prediction filters and residual values |
| * read from the data stream, and update the filter state. */ |
| |
| static void filter_channel(MLPDecodeContext *m, unsigned int substr, |
| unsigned int channel) |
| { |
| SubStream *s = &m->substream[substr]; |
| const int32_t *fircoeff = s->channel_params[channel].coeff[FIR]; |
| int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER]; |
| int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE; |
| int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE; |
| FilterParams *fir = &s->channel_params[channel].filter_params[FIR]; |
| FilterParams *iir = &s->channel_params[channel].filter_params[IIR]; |
| unsigned int filter_shift = fir->shift; |
| int32_t mask = MSB_MASK(s->quant_step_size[channel]); |
| |
| memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t)); |
| memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t)); |
| |
| m->dsp.mlp_filter_channel(firbuf, fircoeff, |
| fir->order, iir->order, |
| filter_shift, mask, s->blocksize, |
| &m->sample_buffer[s->blockpos][channel]); |
| |
| memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t)); |
| memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t)); |
| } |
| |
| /** Read a block of PCM residual data (or actual if no filtering active). */ |
| |
| static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, |
| unsigned int substr) |
| { |
| SubStream *s = &m->substream[substr]; |
| unsigned int i, ch, expected_stream_pos = 0; |
| int ret; |
| |
| if (s->data_check_present) { |
| expected_stream_pos = get_bits_count(gbp); |
| expected_stream_pos += get_bits(gbp, 16); |
| avpriv_request_sample(m->avctx, |
| "Substreams with VLC block size check info"); |
| } |
| |
| if (s->blockpos + s->blocksize > m->access_unit_size) { |
| av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| memset(&m->bypassed_lsbs[s->blockpos][0], 0, |
| s->blocksize * sizeof(m->bypassed_lsbs[0])); |
| |
| for (i = 0; i < s->blocksize; i++) |
| if ((ret = read_huff_channels(m, gbp, substr, i)) < 0) |
| return ret; |
| |
| for (ch = s->min_channel; ch <= s->max_channel; ch++) |
| filter_channel(m, substr, ch); |
| |
| s->blockpos += s->blocksize; |
| |
| if (s->data_check_present) { |
| if (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; |
| } |
| |
| /** Write the audio data into the output buffer. */ |
| |
| static int output_data(MLPDecodeContext *m, unsigned int substr, |
| AVFrame *frame, int *got_frame_ptr) |
| { |
| AVCodecContext *avctx = m->avctx; |
| SubStream *s = &m->substream[substr]; |
| unsigned int mat; |
| unsigned int maxchan; |
| int ret; |
| int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); |
| |
| if (m->avctx->channels != s->max_matrix_channel + 1) { |
| av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (!s->blockpos) { |
| av_log(avctx, AV_LOG_ERROR, "No samples to output.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| maxchan = s->max_matrix_channel; |
| if (!s->noise_type) { |
| generate_2_noise_channels(m, substr); |
| maxchan += 2; |
| } else { |
| fill_noise_buffer(m, substr); |
| } |
| |
| /* Apply the channel matrices in turn to reconstruct the original audio |
| * samples. */ |
| for (mat = 0; mat < s->num_primitive_matrices; mat++) { |
| unsigned int dest_ch = s->matrix_out_ch[mat]; |
| m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0], |
| s->matrix_coeff[mat], |
| &m->bypassed_lsbs[0][mat], |
| m->noise_buffer, |
| s->num_primitive_matrices - mat, |
| dest_ch, |
| s->blockpos, |
| maxchan, |
| s->matrix_noise_shift[mat], |
| m->access_unit_size_pow2, |
| MSB_MASK(s->quant_step_size[dest_ch])); |
| } |
| |
| /* get output buffer */ |
| frame->nb_samples = s->blockpos; |
| if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| return ret; |
| s->lossless_check_data = m->dsp.mlp_pack_output(s->lossless_check_data, |
| s->blockpos, |
| m->sample_buffer, |
| frame->data[0], |
| s->ch_assign, |
| s->output_shift, |
| s->max_matrix_channel, |
| is32); |
| |
| /* Update matrix encoding side data */ |
| if ((ret = ff_side_data_update_matrix_encoding(frame, s->matrix_encoding)) < 0) |
| return ret; |
| |
| *got_frame_ptr = 1; |
| |
| return 0; |
| } |
| |
| /** Read an access unit from the stream. |
| * @return negative on error, 0 if not enough data is present in the input stream, |
| * otherwise the number of bytes consumed. */ |
| |
| static int read_access_unit(AVCodecContext *avctx, void* data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| const uint8_t *buf = avpkt->data; |
| int buf_size = avpkt->size; |
| MLPDecodeContext *m = avctx->priv_data; |
| 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; |
| int ret; |
| |
| if (buf_size < 4) |
| return AVERROR_INVALIDDATA; |
| |
| length = (AV_RB16(buf) & 0xfff) * 2; |
| |
| if (length < 4 || length > buf_size) |
| return AVERROR_INVALIDDATA; |
| |
| init_get_bits(&gb, (buf + 4), (length - 4) * 8); |
| |
| m->is_major_sync_unit = 0; |
| if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) { |
| if (read_major_sync(m, &gb) < 0) |
| goto error; |
| m->is_major_sync_unit = 1; |
| header_size += m->major_sync_header_size; |
| } |
| |
| if (!m->params_valid) { |
| av_log(m->avctx, AV_LOG_WARNING, |
| "Stream parameters not seen; skipping frame.\n"); |
| *got_frame_ptr = 0; |
| return length; |
| } |
| |
| substream_start = 0; |
| |
| for (substr = 0; substr < m->num_substreams; substr++) { |
| int extraword_present, checkdata_present, end, nonrestart_substr; |
| |
| extraword_present = get_bits1(&gb); |
| nonrestart_substr = get_bits1(&gb); |
| checkdata_present = get_bits1(&gb); |
| skip_bits1(&gb); |
| |
| end = get_bits(&gb, 12) * 2; |
| |
| substr_header_size += 2; |
| |
| if (extraword_present) { |
| if (m->avctx->codec_id == AV_CODEC_ID_MLP) { |
| av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n"); |
| goto error; |
| } |
| skip_bits(&gb, 16); |
| substr_header_size += 2; |
| } |
| |
| if (!(nonrestart_substr ^ m->is_major_sync_unit)) { |
| av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n"); |
| goto error; |
| } |
| |
| if (end + header_size + substr_header_size > length) { |
| av_log(m->avctx, AV_LOG_ERROR, |
| "Indicated length of substream %d data goes off end of " |
| "packet.\n", substr); |
| end = length - header_size - substr_header_size; |
| } |
| |
| if (end < substream_start) { |
| av_log(avctx, AV_LOG_ERROR, |
| "Indicated end offset of substream %d data " |
| "is smaller than calculated start offset.\n", |
| substr); |
| goto error; |
| } |
| |
| if (substr > m->max_decoded_substream) |
| continue; |
| |
| substream_parity_present[substr] = checkdata_present; |
| substream_data_len[substr] = end - substream_start; |
| substream_start = end; |
| } |
| |
| parity_bits = ff_mlp_calculate_parity(buf, 4); |
| parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size); |
| |
| if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) { |
| av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n"); |
| goto error; |
| } |
| |
| buf += header_size + substr_header_size; |
| |
| for (substr = 0; substr <= m->max_decoded_substream; substr++) { |
| SubStream *s = &m->substream[substr]; |
| init_get_bits(&gb, buf, substream_data_len[substr] * 8); |
| |
| m->matrix_changed = 0; |
| memset(m->filter_changed, 0, sizeof(m->filter_changed)); |
| |
| 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) |
| goto next_substr; |
| if (read_decoding_params(m, &gb, substr) < 0) |
| goto next_substr; |
| } |
| |
| if (!s->restart_seen) |
| goto next_substr; |
| |
| if ((ret = read_block_data(m, &gb, substr)) < 0) |
| return ret; |
| |
| if (get_bits_count(&gb) >= substream_data_len[substr] * 8) |
| goto substream_length_mismatch; |
| |
| } while (!get_bits1(&gb)); |
| |
| skip_bits(&gb, (-get_bits_count(&gb)) & 15); |
| |
| if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) { |
| int shorten_by; |
| |
| if (get_bits(&gb, 16) != 0xD234) |
| return AVERROR_INVALIDDATA; |
| |
| shorten_by = get_bits(&gb, 16); |
| if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by & 0x2000) |
| s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos); |
| else if (m->avctx->codec_id == AV_CODEC_ID_MLP && shorten_by != 0xD234) |
| return AVERROR_INVALIDDATA; |
| |
| if (substr == m->max_decoded_substream) |
| av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n"); |
| } |
| |
| if (substream_parity_present[substr]) { |
| uint8_t parity, checksum; |
| |
| if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16) |
| goto substream_length_mismatch; |
| |
| parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2); |
| checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2); |
| |
| if ((get_bits(&gb, 8) ^ parity) != 0xa9 ) |
| av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr); |
| if ( get_bits(&gb, 8) != checksum) |
| av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr); |
| } |
| |
| if (substream_data_len[substr] * 8 != get_bits_count(&gb)) |
| goto substream_length_mismatch; |
| |
| next_substr: |
| if (!s->restart_seen) |
| av_log(m->avctx, AV_LOG_ERROR, |
| "No restart header present in substream %d.\n", substr); |
| |
| buf += substream_data_len[substr]; |
| } |
| |
| if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0) |
| return ret; |
| |
| return length; |
| |
| substream_length_mismatch: |
| av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr); |
| return AVERROR_INVALIDDATA; |
| |
| error: |
| m->params_valid = 0; |
| return AVERROR_INVALIDDATA; |
| } |
| |
| #if CONFIG_MLP_DECODER |
| AVCodec ff_mlp_decoder = { |
| .name = "mlp", |
| .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_MLP, |
| .priv_data_size = sizeof(MLPDecodeContext), |
| .init = mlp_decode_init, |
| .decode = read_access_unit, |
| .capabilities = AV_CODEC_CAP_DR1, |
| }; |
| #endif |
| #if CONFIG_TRUEHD_DECODER |
| AVCodec ff_truehd_decoder = { |
| .name = "truehd", |
| .long_name = NULL_IF_CONFIG_SMALL("TrueHD"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_TRUEHD, |
| .priv_data_size = sizeof(MLPDecodeContext), |
| .init = mlp_decode_init, |
| .decode = read_access_unit, |
| .capabilities = AV_CODEC_CAP_DR1, |
| }; |
| #endif /* CONFIG_TRUEHD_DECODER */ |
