44 "Please file a bug report following the instructions at "
45 "http://libav.org/bugreports.html and include "
46 "a sample of this file.";
79 #define PARAM_BLOCKSIZE (1 << 7)
80 #define PARAM_MATRIX (1 << 6)
81 #define PARAM_OUTSHIFT (1 << 5)
82 #define PARAM_QUANTSTEP (1 << 4)
83 #define PARAM_FIR (1 << 3)
84 #define PARAM_IIR (1 << 2)
85 #define PARAM_HUFFOFFSET (1 << 1)
86 #define PARAM_PRESENCE (1 << 0)
161 if (!huff_vlc[0].
bits) {
177 unsigned int substr,
unsigned int ch)
186 sign_huff_offset -= 7 << lsb_bits;
189 sign_huff_offset -= 1 << sign_shift;
191 return sign_huff_offset;
198 unsigned int substr,
unsigned int pos)
201 unsigned int mat, channel;
207 for (channel = s->
min_channel; channel <= s->max_channel; channel++) {
211 int lsb_bits = cp->
huff_lsbs - quant_step_size;
215 result =
get_vlc2(gbp, huff_vlc[codebook-1].table,
222 result = (result << lsb_bits) +
get_bits(gbp, lsb_bits);
225 result <<= quant_step_size;
269 "Channel group 2 cannot have more bits per sample than group 1.\n");
275 "Channel groups with differing sample rates are not currently supported.\n");
285 "Sampling rate %d is greater than the supported maximum (%d).\n",
291 "Block size %d is greater than the supported maximum (%d).\n",
297 "Block size pow2 %d is greater than the supported maximum (%d).\n",
310 "Number of substreams %d is larger than the maximum supported "
342 const uint8_t *buf,
unsigned int substr)
348 uint8_t lossless_check;
350 int min_channel, max_channel, max_matrix_channel;
357 if (sync_word != 0x31ea >> 1) {
359 "restart header sync incorrect (got 0x%04x)\n", sync_word);
374 max_matrix_channel =
get_bits(gbp, 4);
376 if (max_matrix_channel > std_max_matrix_channel) {
378 "Max matrix channel cannot be greater than %d.\n",
383 if (max_channel != max_matrix_channel) {
385 "Max channel must be equal max matrix channel.\n");
393 "Number of channels %d is larger than the maximum supported "
398 if (min_channel > max_channel) {
400 "Substream min channel cannot be greater than max channel.\n");
409 if (m->
avctx->request_channels > 0 &&
413 "Extracting %d channel downmix from substream %d. "
414 "Further substreams will be skipped.\n",
429 if (tmp != lossless_check)
431 "Lossless check failed - expected %02x, calculated %02x.\n",
432 lossless_check, tmp);
443 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
464 for (ch = s->
min_channel; ch <= s->max_channel; ch++) {
487 unsigned int substr,
unsigned int channel,
493 const char fchar = filter ?
'I' :
'F';
505 if (order > max_order) {
507 "%cIR filter order %d is greater than maximum %d.\n",
508 fchar, order, max_order);
515 int coeff_bits, coeff_shift;
521 if (coeff_bits < 1 || coeff_bits > 16) {
523 "%cIR filter coeff_bits must be between 1 and 16.\n",
527 if (coeff_bits + coeff_shift > 16) {
529 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
534 for (i = 0; i < order; i++)
535 fcoeff[i] =
get_sbits(gbp, coeff_bits) << coeff_shift;
538 int state_bits, state_shift;
542 "FIR filter has state data specified.\n");
551 for (i = 0; i < order; i++)
564 unsigned int mat, ch;
578 "Number of primitive matrices cannot be greater than %d.\n",
579 max_primitive_matrices);
584 int frac_bits, max_chan;
591 "Invalid channel %d specified as output from matrix.\n",
595 if (frac_bits > 14) {
597 "Too many fractional bits specified.\n");
605 for (ch = 0; ch <= max_chan; ch++) {
608 coeff_val =
get_sbits(gbp, frac_bits + 2);
610 s->
matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
651 "FIR and IIR filters must use the same precision.\n");
723 for (ch = s->
min_channel; ch <= s->max_channel; ch++)
731 #define MSB_MASK(bits) (-1u << bits)
737 unsigned int channel)
746 unsigned int filter_shift = fir->
shift;
767 unsigned int i, ch, expected_stream_pos = 0;
772 expected_stream_pos +=
get_bits(gbp, 16);
789 for (ch = s->
min_channel; ch <= s->max_channel; ch++)
806 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
807 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
808 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
809 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
810 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
811 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
812 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
813 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
814 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
815 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
816 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
817 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
818 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
819 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
820 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
821 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
842 uint16_t seed_shr7 = seed >> 7;
846 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
861 uint8_t seed_shr15 = seed >> 15;
863 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
876 unsigned int mat, src_ch, i;
877 unsigned int maxchan;
893 int index2 = 2 * index + 1;
902 for (src_ch = 0; src_ch <= maxchan; src_ch++)
903 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
905 if (matrix_noise_shift) {
911 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
919 void *
data,
int *got_frame_ptr)
923 unsigned int i, out_ch = 0;
949 if (is32) *data_32++ = sample << 8;
950 else *data_16++ = sample >> 8;
965 int *got_frame_ptr,
AVPacket *avpkt)
967 const uint8_t *buf = avpkt->
data;
968 int buf_size = avpkt->
size;
971 unsigned int length, substr;
972 unsigned int substream_start;
973 unsigned int header_size = 4;
974 unsigned int substr_header_size = 0;
983 length = (
AV_RB16(buf) & 0xfff) * 2;
985 if (length < 4 || length > buf_size)
1000 "Stream parameters not seen; skipping frame.\n");
1005 substream_start = 0;
1008 int extraword_present, checkdata_present, end, nonrestart_substr;
1017 substr_header_size += 2;
1019 if (extraword_present) {
1025 substr_header_size += 2;
1033 if (end + header_size + substr_header_size > length) {
1035 "Indicated length of substream %d data goes off end of "
1036 "packet.\n", substr);
1037 end = length - header_size - substr_header_size;
1040 if (end < substream_start) {
1042 "Indicated end offset of substream %d data "
1043 "is smaller than calculated start offset.\n",
1051 substream_parity_present[substr] = checkdata_present;
1052 substream_data_len[substr] = end - substream_start;
1053 substream_start = end;
1059 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1064 buf += header_size + substr_header_size;
1096 goto substream_length_mismatch;
1102 if (substream_data_len[substr] * 8 -
get_bits_count(&gb) >= 32) {
1118 if (substream_parity_present[substr]) {
1119 uint8_t parity, checksum;
1122 goto substream_length_mismatch;
1127 if ((
get_bits(&gb, 8) ^ parity) != 0xa9 )
1134 goto substream_length_mismatch;
1139 "No restart header present in substream %d.\n", substr);
1141 buf += substream_data_len[substr];
1151 substream_length_mismatch:
1171 #if CONFIG_TRUEHD_DECODER
uint8_t shift
Right shift to apply to output of filter.
static unsigned int show_bits_long(GetBitContext *s, int n)
Show 0-32 bits.
static const int16_t coeffs[28]
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
struct SubStream SubStream
AV_WL32 AV_WL24 AV_WL16 AV_WB32 AV_WB24 AV_RB16
FilterParams filter_params[NUM_FILTERS]
static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr)
Read decoding parameters that change more often than those in the restart header. ...
int8_t noise_buffer[MAX_BLOCKSIZE_POW2]
uint8_t param_presence_flags
Bitmask of which parameter sets are conveyed in a decoding parameter block.
static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
Noise generation functions.
uint8_t params_valid
Set if a valid major sync block has been read. Otherwise no decoding is possible. ...
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
AVFrame * coded_frame
the picture in the bitstream
#define AV_LOG_WARNING
Something somehow does not look correct.
struct MLPDecodeContext MLPDecodeContext
const uint8_t ff_mlp_huffman_tables[3][18][2]
Tables defining the Huffman codes.
#define MAX_BLOCKSIZE_POW2
next power of two greater than MAX_BLOCKSIZE
#define MAX_SAMPLERATE
maximum sample frequency seen in files
int8_t output_shift[MAX_CHANNELS]
Left shift to apply to decoded PCM values to get final 24-bit output.
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
static int get_sbits(GetBitContext *s, int n)
int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS]
Matrix coefficients, stored as 2.14 fixed point.
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
enum AVSampleFormat sample_fmt
audio sample format
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
static uint8_t xor_32_to_8(uint32_t value)
XOR four bytes into one.
#define MAX_FIR_ORDER
The maximum number of taps in IIR and FIR filters.
uint8_t ch_assign[MAX_CHANNELS]
For each channel output by the matrix, the output channel to map it to.
static int get_bits_count(const GetBitContext *s)
static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
Generate a block of noise, used when restart sync word == 0x31eb.
uint8_t restart_seen
Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
bitstream reader API header.
uint8_t min_channel
The index of the first channel coded in this substream.
static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr, unsigned int channel, unsigned int filter)
Read parameters for one of the prediction filters.
static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, const uint8_t *buf, unsigned int substr)
Read a restart header from a block in a substream.
static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
Read parameters for primitive matrices.
av_cold void dsputil_init(DSPContext *c, AVCodecContext *avctx)
static int init(AVCodecParserContext *s)
static int read_access_unit(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Read an access unit from the stream.
int16_t huff_offset
Offset to apply to residual values.
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
#define NUM_FILTERS
number of allowed filters
uint8_t max_channel
The index of the last channel coded in this substream.
uint8_t ff_mlp_calculate_parity(const uint8_t *buf, unsigned int buf_size)
XOR together all the bytes of a buffer.
ChannelParams channel_params[MAX_CHANNELS]
Channel coding parameters for channels in the substream.
#define MAX_MATRIX_CHANNEL_TRUEHD
static const uint16_t mask[17]
int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]
uint8_t quant_step_size[MAX_CHANNELS]
Left shift to apply to Huffman-decoded residuals.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void av_log(void *avcl, int level, const char *fmt,...)
const char * name
Name of the codec implementation.
uint8_t num_substreams
Number of substreams contained within this stream.
uint8_t max_matrix_channel
The number of channels input into the rematrix stage.
static av_cold int mlp_decode_init(AVCodecContext *avctx)
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
#define MAX_MATRIX_CHANNEL_MLP
Last possible matrix channel for each codec.
uint8_t ff_mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
Calculate an 8-bit checksum over a restart header – a non-multiple-of-8 number of bits...
uint16_t noise_type
restart header data
static int read_channel_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp, unsigned int ch)
Read channel parameters.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame)
int32_t lossless_check_data
Running XOR of all output samples.
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
int filter_changed[MAX_CHANNELS][NUM_FILTERS]
uint8_t lsb_bypass[MAX_MATRICES]
Whether the LSBs of the matrix output are encoded in the bitstream.
int32_t coeff[NUM_FILTERS][MAX_FIR_ORDER]
static int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr, unsigned int pos)
Read a sample, consisting of either, both or neither of entropy-coded MSBs and plain LSBs...
int access_unit_size
number of PCM samples contained in each frame
static int output_data(MLPDecodeContext *m, unsigned int substr, void *data, int *got_frame_ptr)
Write the audio data into the output buffer.
static int32_t calculate_sign_huff(MLPDecodeContext *m, unsigned int substr, unsigned int ch)
int frame_size
Samples per packet, initialized when calling 'init'.
uint16_t blocksize
number of PCM samples in current audio block
uint8_t codebook
Which VLC codebook to use to read residuals.
#define MAX_MATRICES_TRUEHD
#define MAX_BLOCKSIZE
maximum number of audio samples within one access unit
uint8_t data_check_present
Set if the substream contains extra info to check the size of VLC blocks.
int32_t state[MAX_FIR_ORDER]
int sample_rate
samples per second
#define VLC_BITS
number of bits used for VLC lookup - longest Huffman code is 9
SubStream substream[MAX_SUBSTREAMS]
uint8_t order
number of taps in filter
main external API structure.
int is_major_sync_unit
Current access unit being read has a major sync.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS]
static unsigned int get_bits1(GetBitContext *s)
static void skip_bits1(GetBitContext *s)
static void skip_bits(GetBitContext *s, int n)
static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
Apply the channel matrices in turn to reconstruct the original audio samples.
static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr)
Read a block of PCM residual data (or actual if no filtering active).
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
uint8_t num_primitive_matrices
matrix data
uint8_t max_decoded_substream
Index of the last substream to decode - further substreams are skipped.
static const int8_t noise_table[256]
Data table used for TrueHD noise generation function.
static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
Read a major sync info header - contains high level information about the stream - sample rate...
uint8_t huff_lsbs
Size of residual suffix not encoded using VLC.
uint16_t blockpos
Number of PCM samples decoded so far in this frame.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static av_cold void init_static(void)
Initialize static data, constant between all invocations of the codec.
uint32_t noisegen_seed
The current seed value for the pseudorandom noise generator(s).
common internal api header.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
uint8_t matrix_out_ch[MAX_MATRICES]
matrix output channel
int access_unit_size_pow2
next power of two above the number of samples in each frame
#define MAX_SUBSTREAMS
Maximum number of substreams that can be decoded.
int ff_mlp_read_major_sync(void *log, MLPHeaderInfo *mh, GetBitContext *gb)
Read a major sync info header - contains high level information about the stream - sample rate...
uint8_t matrix_noise_shift[MAX_MATRICES]
Left shift to apply to noise values in 0x31eb substreams.
uint8_t noise_shift
The left shift applied to random noise in 0x31ea substreams.
static void filter_channel(MLPDecodeContext *m, unsigned int substr, unsigned int channel)
Generate PCM samples using the prediction filters and residual values read from the data stream...
sample data coding information
int channels
number of audio channels
av_cold void ff_mlp_init_crc(void)
void(* mlp_filter_channel)(int32_t *state, const int32_t *coeff, int firorder, int iirorder, unsigned int filter_shift, int32_t mask, int blocksize, int32_t *sample_buffer)
#define MAX_MATRICES_MLP
Maximum number of matrices used in decoding; most streams have one matrix per output channel...
void avcodec_get_frame_defaults(AVFrame *pic)
Set the fields of the given AVFrame to default values.
int nb_samples
number of audio samples (per channel) described by this frame
static const char * sample_message
int32_t sign_huff_offset
sign/rounding-corrected version of huff_offset
uint8_t ff_mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
MLP uses checksums that seem to be based on the standard CRC algorithm, but are not (in implementatio...