42 #define BITSTREAM_READER_LE
53 #define QDM2_LIST_ADD(list, size, packet) \
56 list[size - 1].next = &list[size]; \
58 list[size].packet = packet; \
59 list[size].next = NULL; \
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
66 #define FIX_NOISE_IDX(noise_idx) \
67 if ((noise_idx) >= 3840) \
68 (noise_idx) -= 3840; \
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
72 #define SAMPLES_NEEDED \
73 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
75 #define SAMPLES_NEEDED_2(why) \
76 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
78 #define QDM2_MAX_FRAME_SIZE 512
200 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
224 if ((
value & ~3) > 0)
252 for (
i = 0;
i < length;
i++)
255 return (uint16_t)(
value & 0xffff);
269 if (sub_packet->
type == 0) {
270 sub_packet->
size = 0;
275 if (sub_packet->
type & 0x80) {
276 sub_packet->
size <<= 8;
278 sub_packet->
type &= 0x7f;
281 if (sub_packet->
type == 0x7f)
302 while (list && list->
packet) {
318 int i, j, n, ch, sum;
323 for (
i = 0;
i < n;
i++) {
326 for (j = 0; j < 8; j++)
333 for (j = 0; j < 8; j++)
355 for (j = 0; j < 64; j++) {
380 for (j = 0; j < 64; ) {
381 if (coding_method[ch][sb][j] < 8)
383 if ((coding_method[ch][sb][j] - 8) > 22) {
387 switch (
switchtable[coding_method[ch][sb][j] - 8]) {
411 for (k = 0; k <
run; k++) {
413 int sbjk = sb + (j + k) / 64;
418 if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
422 memset(&coding_method[ch][sb][j + k], case_val,
424 memset(&coding_method[ch][sb][j + k], case_val,
445 int i, sb, ch, sb_used;
449 for (sb = 0; sb < 30; sb++)
450 for (
i = 0;
i < 8;
i++) {
464 for (sb = 0; sb < sb_used; sb++)
466 for (
i = 0;
i < 64;
i++) {
475 for (sb = 0; sb < sb_used; sb++) {
476 if ((sb >= 4) && (sb <= 23)) {
478 for (
i = 0;
i < 64;
i++) {
492 for (
i = 0;
i < 64;
i++) {
504 for (
i = 0;
i < 64;
i++) {
536 int c,
int superblocktype_2_3,
541 int add1, add2, add3, add4;
544 if (!superblocktype_2_3) {
549 for (sb = 0; sb < 30; sb++) {
550 for (j = 1; j < 63; j++) {
551 add1 = tone_level_idx[ch][sb][j] - 10;
554 add2 = add3 = add4 = 0;
570 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
573 tone_level_idx_temp[ch][sb][j + 1] =
tmp & 0xff;
575 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
580 for (sb = 0; sb < 30; sb++)
581 for (j = 0; j < 64; j++)
582 acc += tone_level_idx_temp[ch][sb][j];
584 multres = 0x66666667LL * (
acc * 10);
585 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
587 for (sb = 0; sb < 30; sb++)
588 for (j = 0; j < 64; j++) {
589 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
620 coding_method[ch][sb][j] = ((
tmp & 0xfffa) + 30 )& 0xff;
622 for (sb = 0; sb < 30; sb++)
625 for (sb = 0; sb < 30; sb++)
626 for (j = 0; j < 64; j++)
628 if (coding_method[ch][sb][j] < 10)
629 coding_method[ch][sb][j] = 10;
632 if (coding_method[ch][sb][j] < 16)
633 coding_method[ch][sb][j] = 16;
635 if (coding_method[ch][sb][j] < 30)
636 coding_method[ch][sb][j] = 30;
641 for (sb = 0; sb < 30; sb++)
642 for (j = 0; j < 64; j++)
660 int length,
int sb_min,
int sb_max)
663 int joined_stereo, zero_encoding;
665 float type34_div = 0;
666 float type34_predictor;
668 int sign_bits[16] = {0};
672 for (sb=sb_min; sb < sb_max; sb++)
678 for (sb = sb_min; sb < sb_max; sb++) {
690 for (j = 0; j < 16; j++)
693 for (j = 0; j < 64; j++)
709 type34_predictor = 0.0;
712 for (j = 0; j < 128; ) {
717 for (k = 0; k < 5; k++) {
718 if ((j + 2 * k) >= 128)
729 for (k = 0; k < 5; k++)
732 for (k = 0; k < 5; k++)
735 for (k = 0; k < 10; k++)
747 f -=
noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
758 for (k = 0; k < 5; k++) {
770 for (k = 0; k < 5; k++)
774 for (k = 0; k < 5; k++)
788 for (k = 0; k < 3; k++)
791 for (k = 0; k < 3; k++)
814 type34_div = (float)(1 <<
get_bits(gb, 2));
815 samples[0] = ((float)
get_bits(gb, 5) - 16.0) / 15.0;
816 type34_predictor = samples[0];
825 type34_predictor = samples[0];
840 for (k = 0; k <
run && j + k < 128; k++) {
842 q->
tone_level[0][sb][(j + k) / 2] * samples[k];
844 if (sign_bits[(j + k) / 8])
846 q->
tone_level[1][sb][(j + k) / 2] * -samples[k];
849 q->
tone_level[1][sb][(j + k) / 2] * samples[k];
853 for (k = 0; k <
run; k++)
884 quantized_coeffs[0] =
level;
886 for (
i = 0;
i < 7; ) {
898 for (k = 1; k <=
run; k++)
931 for (sb = 0; sb < n; sb++)
933 for (j = 0; j < 8; j++) {
937 for (k=0; k < 8; k++) {
943 for (k=0; k < 8; k++)
950 for (sb = 0; sb < n; sb++)
958 for (j = 0; j < 8; j++)
964 for (sb = 0; sb < n; sb++)
966 for (j = 0; j < 8; j++) {
988 for (
i = 1;
i < n;
i++)
993 for (j = 0; j < (8 - 1); ) {
1000 for (k = 1; k <=
run; k++)
1009 for (
i = 0;
i < 8;
i++)
1103 if (nodes[0] && nodes[1] && nodes[2])
1109 if (nodes[0] && nodes[1] && nodes[3])
1124 int i, packet_bytes, sub_packet_size, sub_packets_D;
1125 unsigned int next_index = 0;
1166 for (
i = 0;
i < 6;
i++)
1170 for (
i = 0; packet_bytes > 0;
i++) {
1187 if (next_index >=
header.size)
1195 sub_packet_size = ((packet->
size > 0xff) ? 1 : 0) + packet->
size + 2;
1197 if (packet->
type == 0)
1200 if (sub_packet_size > packet_bytes) {
1201 if (packet->
type != 10 && packet->
type != 11 && packet->
type != 12)
1203 packet->
size += packet_bytes - sub_packet_size;
1206 packet_bytes -= sub_packet_size;
1212 if (packet->
type == 8) {
1215 }
else if (packet->
type >= 9 && packet->
type <= 12) {
1218 }
else if (packet->
type == 13) {
1219 for (j = 0; j < 6; j++)
1221 }
else if (packet->
type == 14) {
1222 for (j = 0; j < 6; j++)
1224 }
else if (packet->
type == 15) {
1227 }
else if (packet->
type >= 16 && packet->
type < 48 &&
1252 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1264 int local_int_4, local_int_8, stereo_phase, local_int_10;
1265 int local_int_14, stereo_exp, local_int_20, local_int_28;
1279 if(local_int_4 < q->group_size)
1285 local_int_4 += local_int_10;
1286 local_int_28 += (1 << local_int_8);
1288 local_int_4 += 8 * local_int_10;
1289 local_int_28 += (8 << local_int_8);
1294 if (local_int_10 <= 2) {
1299 while (
offset >= (local_int_10 - 1)) {
1300 offset += (1 - (local_int_10 - 1));
1301 local_int_4 += local_int_10;
1302 local_int_28 += (1 << local_int_8);
1309 local_int_14 = (
offset >> local_int_8);
1332 if (stereo_phase < 0)
1337 int sub_packet = (local_int_20 + local_int_28);
1347 stereo_exp, stereo_phase);
1363 for (
i = 0;
i < 5;
i++)
1386 (packet->
type < 16 || packet->
type >= 48 ||
1405 }
else if (
type == 31) {
1406 for (j = 0; j < 4; j++)
1408 }
else if (
type == 46) {
1409 for (j = 0; j < 6; j++)
1411 for (j = 0; j < 4; j++)
1417 for (
i = 0, j = -1;
i < 5;
i++)
1432 const double iscale = 2.0 *
M_PI / 512.0;
1454 for (
i = 0;
i < 2;
i++) {
1460 for (
i = 0;
i < 4;
i++) {
1476 const double iscale = 0.25 *
M_PI;
1478 for (ch = 0; ch < q->
channels; ch++) {
1510 for (
i = 0;
i < 4;
i++)
1523 if (offset < q->frequency_range) {
1566 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1571 for (ch = 0; ch < q->
channels; ch++)
1572 for (
i = 0;
i < 8;
i++)
1573 for (k = sb_used; k <
SBLIMIT; k++)
1577 float *samples_ptr = q->
samples + ch;
1579 for (
i = 0;
i < 8;
i++) {
1592 for (ch = 0; ch < q->
channels; ch++)
1661 if (bytestream2_peek_be64(&gb) == (((uint64_t)
MKBETAG(
'f',
'r',
'm',
'a') << 32) |
1662 (uint64_t)
MKBETAG(
'Q',
'D',
'M',
'2')))
1674 size = bytestream2_get_be32(&gb);
1683 if (bytestream2_get_be32(&gb) !=
MKBETAG(
'Q',
'D',
'C',
'A')) {
1690 avctx->
channels =
s->nb_channels =
s->channels = bytestream2_get_be32(&gb);
1699 avctx->
bit_rate = bytestream2_get_be32(&gb);
1700 s->group_size = bytestream2_get_be32(&gb);
1701 s->fft_size = bytestream2_get_be32(&gb);
1702 s->checksum_size = bytestream2_get_be32(&gb);
1703 if (
s->checksum_size >= 1U << 28 ||
s->checksum_size <= 1) {
1708 s->fft_order =
av_log2(
s->fft_size) + 1;
1711 if ((
s->fft_order < 7) || (
s->fft_order > 9)) {
1717 s->group_order =
av_log2(
s->group_size) + 1;
1718 s->frame_size =
s->group_size / 16;
1723 s->sub_sampling =
s->fft_order - 7;
1724 s->frequency_range = 255 / (1 << (2 -
s->sub_sampling));
1731 switch ((
s->sub_sampling * 2 +
s->channels - 1)) {
1732 case 0:
tmp = 40;
break;
1733 case 1:
tmp = 48;
break;
1734 case 2:
tmp = 56;
break;
1735 case 3:
tmp = 72;
break;
1736 case 4:
tmp = 80;
break;
1737 case 5:
tmp = 100;
break;
1738 default:
tmp=
s->sub_sampling;
break;
1745 s->cm_table_select = tmp_val;
1748 s->coeff_per_sb_select = 0;
1750 s->coeff_per_sb_select = 1;
1752 s->coeff_per_sb_select = 2;
1754 if (
s->fft_size != (1 << (
s->fft_order - 1))) {
1810 for (ch = 0; ch < q->
channels; ch++) {
1841 int *got_frame_ptr,
AVPacket *avpkt)
1845 int buf_size = avpkt->
size;
1852 if(buf_size < s->checksum_size)
1861 for (
i = 0;
i < 16;
i++) {
1864 out +=
s->channels *
s->frame_size;
1869 return s->checksum_size;
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
static av_always_inline int bytestream2_get_bytes_left(GetByteContext *g)
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
static av_always_inline void bytestream2_skip(GetByteContext *g, unsigned int size)
audio channel layout utility functions
#define MKBETAG(a, b, c, d)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static void comp(unsigned char *dst, ptrdiff_t dst_stride, unsigned char *src, ptrdiff_t src_stride, int add)
channel
Use these values when setting the channel map with ebur128_set_channel().
bitstream reader API header.
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
static int get_bits_left(GetBitContext *gb)
static unsigned int get_bits1(GetBitContext *s)
static void skip_bits(GetBitContext *s, int n)
static int get_bits_count(const GetBitContext *s)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
@ AV_SAMPLE_FMT_S16
signed 16 bits
static const uint8_t dequant_table[64]
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static int ff_thread_once(char *control, void(*routine)(void))
mpeg audio declarations for both encoder and decoder.
av_cold void ff_mpadsp_init(MPADSPContext *s)
float ff_mpa_synth_window_float[]
void ff_mpa_synth_init_float(void)
void ff_mpa_synth_filter_float(MPADSPContext *s, float *synth_buf_ptr, int *synth_buf_offset, float *window, int *dither_state, float *samples, ptrdiff_t incr, float *sb_samples)
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 10 if not null, else.
static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
QDM2 checksum.
static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 9, init quantized_coeffs with data from it.
static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 12.
#define QDM2_SB_USED(sub_sampling)
static int fix_coding_method_array(int sb, int channels, sb_int8_array coding_method)
Called while processing data from subpackets 11 and 12.
#define SB_DITHERING_NOISE(sb, noise_idx)
static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
#define SAMPLES_NEEDED_2(why)
#define QDM2_LIST_ADD(list, size, packet)
static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 11.
static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
Related to synthesis filter, process data from packet 10 Init part of quantized_coeffs via function i...
static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static void qdm2_decode_fft_packets(QDM2Context *q)
static void average_quantized_coeffs(QDM2Context *q)
Replace 8 elements with their average value.
#define FIX_NOISE_IDX(noise_idx)
static av_cold void qdm2_init_static_data(void)
Init static data (does not depend on specific file)
static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, GetBitContext *gb)
Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch...
static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
Init parameters from codec extradata.
static void qdm2_fft_decode_tones(QDM2Context *q, int duration, GetBitContext *gb, int b)
#define QDM2_MAX_FRAME_SIZE
static const int switchtable[23]
static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, int offset, int duration, int channel, int exp, int phase)
int8_t sb_int8_array[2][30][64]
static void fill_tone_level_array(QDM2Context *q, int flag)
Related to synthesis filter Called by process_subpacket_10.
static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
static void qdm2_synthesis_filter(QDM2Context *q, int index)
static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
static void qdm2_decode_sub_packet_header(GetBitContext *gb, QDM2SubPacket *sub_packet)
Fill a QDM2SubPacket structure with packet type, size, and data pointer.
static void build_sb_samples_from_noise(QDM2Context *q, int sb)
Build subband samples with noise weighted by q->tone_level.
static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
Process new subpackets for synthesis filter.
static void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, sb_int8_array coding_method, int nb_channels, int c, int superblocktype_2_3, int cm_table_select)
Related to synthesis filter Called by process_subpacket_11 c is built with data from subpacket 11 Mos...
static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8.
static void qdm2_decode_super_block(QDM2Context *q)
Decode superblock, fill packet lists.
static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
static QDM2SubPNode * qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, int type)
Return node pointer to first packet of requested type in list.
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
#define SOFTCLIP_THRESHOLD
static av_cold void init_noise_samples(void)
static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD+1]
static float noise_samples[128]
static uint8_t random_dequant_index[256][5]
static VLC fft_stereo_exp_vlc
static VLC fft_stereo_phase_vlc
static VLC vlc_tab_fft_tone_offset[5]
static VLC vlc_tab_type34
static VLC vlc_tab_tone_level_idx_mid
static VLC fft_level_exp_alt_vlc
static VLC vlc_tab_type30
static av_cold void qdm2_init_vlc(void)
static av_cold void rnd_table_init(void)
static uint8_t random_dequant_type24[128][3]
static av_cold void softclip_table_init(void)
static VLC fft_level_exp_vlc
static VLC vlc_tab_tone_level_idx_hi2
#define HARDCLIP_THRESHOLD
static VLC vlc_tab_tone_level_idx_hi1
static const int16_t fft_level_index_table[256]
static const uint8_t fft_subpackets[32]
static const int vlc_stage3_values[60]
static const int fft_cutoff_index_table[4][2]
static const float fft_tone_envelope_table[4][31]
static const int8_t tone_level_idx_offset_table[30][4]
static const uint8_t coeff_per_sb_for_dequant[3][30]
static const uint8_t coeff_per_sb_for_avg[3][30]
static const float fft_tone_sample_table[4][16][5]
static const float type34_delta[10]
static const float type30_dequant[8]
static const float dequant_1bit[2][3]
static const uint8_t last_coeff[3]
static const float fft_tone_level_table[2][64]
static const int8_t coding_method_table[5][30]
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
av_cold void ff_rdft_end(RDFTContext *s)
static const uint8_t header[24]
#define FF_ARRAY_ELEMS(a)
main external API structure.
enum AVSampleFormat sample_fmt
audio sample format
int64_t bit_rate
the average bitrate
int sample_rate
samples per second
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
int channels
number of audio channels
uint64_t channel_layout
Audio channel layout.
const char * name
Name of the codec implementation.
This structure describes decoded (raw) audio or video data.
int nb_samples
number of audio samples (per channel) described by this frame
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
This structure stores compressed data.
int coeff_per_sb_select
selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
FFTTone fft_tones[1000]
FFT and tones.
int do_synth_filter
used to perform or skip synthesis filter
int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]
int superblocktype_2_3
select fft tables and some algorithm based on superblock type
float output_buffer[QDM2_MAX_FRAME_SIZE *MPA_MAX_CHANNELS *2]
float tone_level[MPA_MAX_CHANNELS][30][64]
Mixed temporary data used in decoding.
int synth_buf_offset[MPA_MAX_CHANNELS]
int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]
int fft_size
size of FFT, in complex numbers
float synth_buf[MPA_MAX_CHANNELS][512 *2]
QDM2SubPNode sub_packet_list_A[16]
list of all packets
int channels
number of channels
int8_t coding_method[MPA_MAX_CHANNELS][30][64]
float sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]
int noise_idx
index for dithering noise table
MPADSPContext mpadsp
Synthesis filter.
float samples[MPA_MAX_CHANNELS *MPA_FRAME_SIZE]
const uint8_t * compressed_data
I/O data.
int fft_coefs_min_index[5]
int sub_packets_B
number of packets on 'B' list
int fft_order
order of FFT (actually fftorder+1)
int group_order
Parameters built from header parameters, do not change during playback.
FFTCoefficient fft_coefs[1000]
int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]
int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]
int frame_size
size of data frame
int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]
int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]
int fft_coefs_max_index[5]
QDM2SubPNode sub_packet_list_D[16]
DCT packets.
int sub_sampling
subsampling: 0=25%, 1=50%, 2=100% */
QDM2SubPNode sub_packet_list_C[16]
packets with errors?
int nb_channels
Parameters from codec header, do not change during playback.
QDM2SubPacket sub_packets[16]
Packets and packet lists.
int cm_table_select
selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
int checksum_size
size of data block, used also for checksum
int has_errors
packet has errors
int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]
QDM2SubPNode sub_packet_list_B[16]
FFT packets B are on list.
int group_size
size of frame group (16 frames per group)
QDM2Complex complex[MPA_MAX_CHANNELS][256]
A node in the subpacket list.
QDM2SubPacket * packet
packet
struct QDM2SubPNode * next
pointer to next packet in the list, NULL if leaf node
const uint8_t * data
pointer to subpacket data (points to input data buffer, it's not a private copy)
unsigned int size
subpacket size
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
VLC_TYPE(* table)[2]
code, bits
#define avpriv_request_sample(...)
static const struct twinvq_data tab
static av_always_inline int diff(const uint32_t a, const uint32_t b)
static const uint8_t offset[127][2]