| /* | |
| * 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 "config_components.h" | |
| #include <stdint.h> | |
| #include "avcodec.h" | |
| #include "libavutil/internal.h" | |
| #include "libavutil/intreadwrite.h" | |
| #include "libavutil/channel_layout.h" | |
| #include "libavutil/mem_internal.h" | |
| #include "libavutil/thread.h" | |
| #include "libavutil/opt.h" | |
| #include "codec_internal.h" | |
| #include "decode.h" | |
| #include "get_bits.h" | |
| #include "mlp_parse.h" | |
| #include "mlpdsp.h" | |
| #include "mlp.h" | |
| #include "config.h" | |
| #include "profiles.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; | |
| /// Set if end of stream is encountered | |
| uint8_t end_of_stream; | |
| //@{ | |
| /** 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 coded channels mask in this substream. | |
| uint64_t coded_channels; | |
| /// The number of channels input into the rematrix stage. | |
| uint8_t max_matrix_channel; | |
| /// For each channel output by the matrix, the output channel to map it to | |
| uint8_t ch_assign[MAX_CHANNELS]; | |
| /// The channel layout for this substream | |
| uint64_t mask; | |
| /// The matrix encoding mode for this substream | |
| enum AVMatrixEncoding matrix_encoding; | |
| enum AVMatrixEncoding prev_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; | |
| //@} | |
| //@{ | |
| /** 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 { | |
| const AVClass *class; | |
| AVCodecContext *avctx; | |
| AVChannelLayout downmix_layout; | |
| /// 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; | |
| /// Which substream of substreams carry 16-channel presentation | |
| uint8_t extended_substream_info; | |
| /// Which substream of substreams carry 2/6/8-channel presentation | |
| uint8_t substream_info; | |
| /// 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; | |
| int32_t (*pack_output)(int32_t lossless_check_data, | |
| uint16_t blockpos, | |
| int32_t (*sample_buffer)[MAX_CHANNELS], | |
| void *data, | |
| uint8_t *ch_assign, | |
| int8_t *output_shift, | |
| uint8_t max_matrix_channel, | |
| int is32); | |
| } MLPDecodeContext; | |
| static const enum AVChannel thd_channel_order[] = { | |
| AV_CHAN_FRONT_LEFT, AV_CHAN_FRONT_RIGHT, // LR | |
| AV_CHAN_FRONT_CENTER, // C | |
| AV_CHAN_LOW_FREQUENCY, // LFE | |
| AV_CHAN_SIDE_LEFT, AV_CHAN_SIDE_RIGHT, // LRs | |
| AV_CHAN_TOP_FRONT_LEFT, AV_CHAN_TOP_FRONT_RIGHT, // LRvh | |
| AV_CHAN_FRONT_LEFT_OF_CENTER, AV_CHAN_FRONT_RIGHT_OF_CENTER, // LRc | |
| AV_CHAN_BACK_LEFT, AV_CHAN_BACK_RIGHT, // LRrs | |
| AV_CHAN_BACK_CENTER, // Cs | |
| AV_CHAN_TOP_CENTER, // Ts | |
| AV_CHAN_SURROUND_DIRECT_LEFT, AV_CHAN_SURROUND_DIRECT_RIGHT, // LRsd | |
| AV_CHAN_WIDE_LEFT, AV_CHAN_WIDE_RIGHT, // LRw | |
| AV_CHAN_TOP_FRONT_CENTER, // Cvh | |
| AV_CHAN_LOW_FREQUENCY_2, // LFE2 | |
| }; | |
| static int mlp_channel_layout_subset(AVChannelLayout *layout, uint64_t mask) | |
| { | |
| return av_channel_layout_check(layout) && | |
| av_channel_layout_subset(layout, mask) == | |
| av_channel_layout_subset(layout, UINT64_MAX); | |
| } | |
| static enum AVChannel thd_channel_layout_extract_channel(uint64_t channel_layout, | |
| int index) | |
| { | |
| int i; | |
| if (av_popcount64(channel_layout) <= index) | |
| return AV_CHAN_NONE; | |
| for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++) | |
| if (channel_layout & (1ULL << thd_channel_order[i]) && !index--) | |
| return thd_channel_order[i]; | |
| return AV_CHAN_NONE; | |
| } | |
| static VLC huff_vlc[3]; | |
| /** Initialize static data, constant between all invocations of the codec. */ | |
| static av_cold void init_static(void) | |
| { | |
| for (int i = 0; i < 3; i++) { | |
| static VLCElem vlc_buf[3 * VLC_STATIC_SIZE]; | |
| huff_vlc[i].table = &vlc_buf[i * VLC_STATIC_SIZE]; | |
| huff_vlc[i].table_allocated = VLC_STATIC_SIZE; | |
| vlc_init(&huff_vlc[i], VLC_BITS, 18, | |
| &ff_mlp_huffman_tables[i][0][1], 2, 1, | |
| &ff_mlp_huffman_tables[i][0][0], 2, 1, VLC_INIT_USE_STATIC); | |
| } | |
| 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_long(gbp, lsb_bits); | |
| result += cp->sign_huff_offset; | |
| result *= 1 << quant_step_size; | |
| m->sample_buffer[pos + s->blockpos][channel] = result; | |
| } | |
| return 0; | |
| } | |
| static av_cold int mlp_decode_init(AVCodecContext *avctx) | |
| { | |
| static AVOnce init_static_once = AV_ONCE_INIT; | |
| MLPDecodeContext *m = avctx->priv_data; | |
| int substr; | |
| m->avctx = avctx; | |
| for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
| m->substream[substr].lossless_check_data = 0xffffffff; | |
| ff_mlpdsp_init(&m->dsp); | |
| if (m->downmix_layout.nb_channels) { | |
| if (!av_channel_layout_compare(&m->downmix_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO) || | |
| !av_channel_layout_compare(&m->downmix_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO_DOWNMIX)) { | |
| av_channel_layout_uninit(&avctx->ch_layout); | |
| avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO; | |
| } else if (!av_channel_layout_compare(&m->downmix_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0)) { | |
| av_channel_layout_uninit(&avctx->ch_layout); | |
| avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0; | |
| } else if (!av_channel_layout_compare(&m->downmix_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1)) { | |
| av_channel_layout_uninit(&avctx->ch_layout); | |
| avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1; | |
| } | |
| else | |
| av_log(avctx, AV_LOG_WARNING, "Invalid downmix layout\n"); | |
| } | |
| ff_thread_once(&init_static_once, init_static); | |
| 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; | |
| m->extended_substream_info = mh.extended_substream_info; | |
| m->substream_info = mh.substream_info; | |
| /* If there is a 4th substream and the MSB of substream_info is set, | |
| * there is a 16-channel spatial presentation (Atmos in TrueHD). | |
| */ | |
| if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD | |
| && m->num_substreams == 4 && m->substream_info >> 7 == 1) { | |
| m->avctx->profile = AV_PROFILE_TRUEHD_ATMOS; | |
| } | |
| /* 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->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 != SYNC_MLP) { | |
| 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].mask = AV_CH_LAYOUT_STEREO; | |
| m->substream[substr].mask = mh.channel_layout_mlp; | |
| } else { | |
| if (mh.stream_type != SYNC_TRUEHD) { | |
| avpriv_request_sample(m->avctx, | |
| "unexpected stream_type %X in !MLP", | |
| mh.stream_type); | |
| return AVERROR_PATCHWELCOME; | |
| } | |
| m->substream[1].mask = mh.channel_layout_thd_stream1; | |
| if (mh.channels_thd_stream1 == 2 && | |
| mh.channels_thd_stream2 == 2 && | |
| m->avctx->ch_layout.nb_channels == 2) | |
| m->substream[0].mask = AV_CH_LAYOUT_STEREO; | |
| if ((substr = (mh.num_substreams > 1))) | |
| m->substream[0].mask = AV_CH_LAYOUT_STEREO; | |
| if (mh.num_substreams == 1 && | |
| mh.channels_thd_stream1 == 1 && | |
| mh.channels_thd_stream2 == 1 && | |
| m->avctx->ch_layout.nb_channels == 1) | |
| m->substream[0].mask = AV_CH_LAYOUT_MONO; | |
| if (mh.num_substreams > 2) | |
| if (mh.channel_layout_thd_stream2) | |
| m->substream[2].mask = mh.channel_layout_thd_stream2; | |
| else | |
| m->substream[2].mask = mh.channel_layout_thd_stream1; | |
| if (m->avctx->ch_layout.nb_channels > 2) | |
| if (mh.num_substreams > 2) | |
| m->substream[1].mask = mh.channel_layout_thd_stream1; | |
| else | |
| m->substream[mh.num_substreams > 1].mask = mh.channel_layout_thd_stream2; | |
| } | |
| 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, noise_type; | |
| 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; | |
| } | |
| noise_type = get_bits1(gbp); | |
| if (m->avctx->codec_id == AV_CODEC_ID_MLP && 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; | |
| } | |
| /* 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_matrix_channel > MAX_MATRIX_CHANNEL_MLP && !noise_type) { | |
| avpriv_request_sample(m->avctx, | |
| "%d channels (more than the " | |
| "maximum supported by the decoder)", | |
| max_channel + 2); | |
| return AVERROR_PATCHWELCOME; | |
| } | |
| if (max_channel + 1 > MAX_CHANNELS || max_channel + 1 < min_channel) | |
| return AVERROR_INVALIDDATA; | |
| s->min_channel = min_channel; | |
| s->max_channel = max_channel; | |
| s->coded_channels = ((1LL << (max_channel - min_channel + 1)) - 1) << min_channel; | |
| s->max_matrix_channel = max_matrix_channel; | |
| s->noise_type = noise_type; | |
| if (mlp_channel_layout_subset(&m->downmix_layout, s->mask) && | |
| m->max_decoded_substream > substr) { | |
| av_log(m->avctx, AV_LOG_DEBUG, | |
| "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. " | |
| "Further substreams will be skipped.\n", | |
| s->max_channel + 1, s->mask, substr); | |
| m->max_decoded_substream = substr; | |
| } | |
| s->noise_shift = 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) { | |
| AVChannelLayout l; | |
| enum AVChannel channel = thd_channel_layout_extract_channel(s->mask, ch_assign); | |
| av_channel_layout_from_mask(&l, s->mask); | |
| ch_assign = av_channel_layout_index_from_channel(&l, 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) { | |
| av_channel_layout_uninit(&m->avctx->ch_layout); | |
| av_channel_layout_from_mask(&m->avctx->ch_layout, s->mask); | |
| m->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 (s->mask == (AV_CH_LAYOUT_QUAD|AV_CH_LOW_FREQUENCY) || | |
| s->mask == 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 (s->mask == 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) * (1 << 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) * (1 << 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); | |
| goto error; | |
| } | |
| 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]); | |
| goto error; | |
| } | |
| if (frac_bits > 14) { | |
| av_log(m->avctx, AV_LOG_ERROR, | |
| "Too many fractional bits specified.\n"); | |
| goto error; | |
| } | |
| 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 * (1 << (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; | |
| error: | |
| s->num_primitive_matrices = 0; | |
| memset(s->matrix_out_ch, 0, sizeof(s->matrix_out_ch)); | |
| return AVERROR_INVALIDDATA; | |
| } | |
| /** 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->codebook > 0 && cp->huff_lsbs > 24) { | |
| av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n"); | |
| cp->huff_lsbs = 0; | |
| return AVERROR_INVALIDDATA; | |
| } | |
| 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 = 0; | |
| unsigned recompute_sho = 0; | |
| 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 (s->output_shift[ch] < 0) { | |
| avpriv_request_sample(m->avctx, "Negative output_shift"); | |
| s->output_shift[ch] = 0; | |
| } | |
| } | |
| if (substr == m->max_decoded_substream) | |
| m->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++) { | |
| s->quant_step_size[ch] = get_bits(gbp, 4); | |
| recompute_sho |= 1<<ch; | |
| } | |
| for (ch = s->min_channel; ch <= s->max_channel; ch++) | |
| if (get_bits1(gbp)) { | |
| recompute_sho |= 1<<ch; | |
| if ((ret = read_channel_params(m, substr, gbp, ch)) < 0) | |
| goto fail; | |
| } | |
| fail: | |
| for (ch = 0; ch <= s->max_channel; ch++) { | |
| if (recompute_sho & (1<<ch)) { | |
| ChannelParams *cp = &s->channel_params[ch]; | |
| if (cp->codebook > 0 && cp->huff_lsbs < s->quant_step_size[ch]) { | |
| if (ret >= 0) { | |
| av_log(m->avctx, AV_LOG_ERROR, "quant_step_size larger than huff_lsbs\n"); | |
| ret = AVERROR_INVALIDDATA; | |
| } | |
| s->quant_step_size[ch] = 0; | |
| } | |
| cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); | |
| } | |
| } | |
| return ret; | |
| } | |
| #define MSB_MASK(bits) (-(1 << (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)) * (1 << s->noise_shift); | |
| m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) * (1 << 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->ch_layout.nb_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->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 (s->matrix_encoding != s->prev_matrix_encoding) { | |
| if ((ret = ff_side_data_update_matrix_encoding(frame, s->matrix_encoding)) < 0) | |
| return ret; | |
| s->prev_matrix_encoding = s->matrix_encoding; | |
| } | |
| *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, AVFrame *frame, | |
| 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 (length < header_size + substr_header_size) { | |
| av_log(m->avctx, AV_LOG_ERROR, "Insufficient data for headers\n"); | |
| goto error; | |
| } | |
| 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 (((avctx->ch_layout.nb_channels == 6 && | |
| ((m->substream_info >> 2) & 0x3) != 0x3) || | |
| (avctx->ch_layout.nb_channels == 8 && | |
| ((m->substream_info >> 4) & 0x7) != 0x7 && | |
| ((m->substream_info >> 4) & 0x7) != 0x6 && | |
| ((m->substream_info >> 4) & 0x7) != 0x3)) && | |
| substr > 0 && substr < m->max_decoded_substream && | |
| (s->min_channel <= m->substream[substr - 1].max_channel)) { | |
| av_log(avctx, AV_LOG_DEBUG, | |
| "Previous substream(%d) channels overlaps current substream(%d) channels, skipping.\n", | |
| substr - 1, substr); | |
| goto next_substr; | |
| } | |
| if (substr != m->max_decoded_substream && | |
| ((s->coded_channels & m->substream[m->max_decoded_substream].coded_channels) != 0)) { | |
| av_log(avctx, AV_LOG_DEBUG, | |
| "Current substream(%d) channels overlaps final substream(%d) channels, skipping.\n", | |
| substr, m->max_decoded_substream); | |
| 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; | |
| av_log(m->avctx, AV_LOG_DEBUG, "End of stream indicated.\n"); | |
| s->end_of_stream = 1; | |
| } | |
| 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, frame, got_frame_ptr)) < 0) | |
| return ret; | |
| for (substr = 0; substr <= m->max_decoded_substream; substr++){ | |
| SubStream *s = &m->substream[substr]; | |
| if (s->end_of_stream) { | |
| s->lossless_check_data = 0xffffffff; | |
| s->end_of_stream = 0; | |
| m->params_valid = 0; | |
| } | |
| } | |
| 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; | |
| } | |
| static void mlp_decode_flush(AVCodecContext *avctx) | |
| { | |
| MLPDecodeContext *m = avctx->priv_data; | |
| m->params_valid = 0; | |
| for (int substr = 0; substr <= m->max_decoded_substream; substr++){ | |
| SubStream *s = &m->substream[substr]; | |
| s->lossless_check_data = 0xffffffff; | |
| s->prev_matrix_encoding = 0; | |
| } | |
| } | |
| #define OFFSET(x) offsetof(MLPDecodeContext, x) | |
| #define FLAGS (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM) | |
| static const AVOption options[] = { | |
| { "downmix", "Request a specific channel layout from the decoder", OFFSET(downmix_layout), | |
| AV_OPT_TYPE_CHLAYOUT, {.str = NULL}, .flags = FLAGS }, | |
| { NULL }, | |
| }; | |
| static const AVClass mlp_decoder_class = { | |
| .class_name = "MLP decoder", | |
| .item_name = av_default_item_name, | |
| .option = options, | |
| .version = LIBAVUTIL_VERSION_INT, | |
| }; | |
| static const AVClass truehd_decoder_class = { | |
| .class_name = "TrueHD decoder", | |
| .item_name = av_default_item_name, | |
| .option = options, | |
| .version = LIBAVUTIL_VERSION_INT, | |
| }; | |
| #if CONFIG_MLP_DECODER | |
| const FFCodec ff_mlp_decoder = { | |
| .p.name = "mlp", | |
| CODEC_LONG_NAME("MLP (Meridian Lossless Packing)"), | |
| .p.type = AVMEDIA_TYPE_AUDIO, | |
| .p.id = AV_CODEC_ID_MLP, | |
| .priv_data_size = sizeof(MLPDecodeContext), | |
| .p.priv_class = &mlp_decoder_class, | |
| .init = mlp_decode_init, | |
| FF_CODEC_DECODE_CB(read_access_unit), | |
| .flush = mlp_decode_flush, | |
| .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, | |
| }; | |
| #endif | |
| #if CONFIG_TRUEHD_DECODER | |
| const FFCodec ff_truehd_decoder = { | |
| .p.name = "truehd", | |
| CODEC_LONG_NAME("TrueHD"), | |
| .p.type = AVMEDIA_TYPE_AUDIO, | |
| .p.id = AV_CODEC_ID_TRUEHD, | |
| .priv_data_size = sizeof(MLPDecodeContext), | |
| .p.priv_class = &truehd_decoder_class, | |
| .init = mlp_decode_init, | |
| FF_CODEC_DECODE_CB(read_access_unit), | |
| .flush = mlp_decode_flush, | |
| .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, | |
| .p.profiles = NULL_IF_CONFIG_SMALL(ff_truehd_profiles), | |
| }; | |
| #endif /* CONFIG_TRUEHD_DECODER */ |