63 5, 6, 7, 8, 9, 10, 11, 12, 14, 16
87 3.162275, 2.818382, 2.511886, 2.238719, 1.995261, 1.778278, 1.584893,
88 1.412536, 1.258924, 1.122018, 1.000000, 0.891251, 0.794328, 0.707946,
89 0.630957, 0.562341, 0.501187, 0.446683, 0.398107, 0.354813, 0.316227,
90 0.281838, 0.251188, 0.223872, 0.199526, 0.177828, 0.158489, 0.141253,
91 0.125892, 0.112201, 0.100000, 0.089125
99 { { 2, 7 }, { 7, 2 }, },
101 { { 2, 7 }, { 7, 2 }, },
102 { { 2, 7 }, { 5, 5 }, { 7, 2 }, },
103 { { 2, 7 }, { 7, 2 }, { 6, 6 }, },
104 { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 8, 8 }, },
105 { { 2, 7 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
106 { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
117 return ((code - (levels >> 1)) * (1 << 24)) / levels;
129 for (i = 0; i < 128; i++) {
137 for (i = 0; i < 32; i++) {
143 for (i = 0; i < 128; i++) {
155 for (i = 0; i < 7; i++) {
159 for (i = 0; i < 15; i++) {
167 for (i = 0; i < 256; i++) {
168 int v = (i >> 5) - ((i >> 7) << 3) - 5;
174 for (i = 0; i < 256; i++) {
175 int v = (i >> 4) - ((i >> 7) << 4) - 4;
352 }
else if (CONFIG_EAC3_DECODER) {
386 downmix_coeffs[0][1] = downmix_coeffs[1][1] = cmix;
390 downmix_coeffs[0][nf] = downmix_coeffs[1][nf] = smix *
LEVEL_MINUS_3DB;
394 downmix_coeffs[0][nf] = downmix_coeffs[1][nf+1] = smix;
400 norm0 += downmix_coeffs[0][i];
401 norm1 += downmix_coeffs[1][i];
403 norm0 = 1.0f / norm0;
404 norm1 = 1.0f / norm1;
406 downmix_coeffs[0][i] *= norm0;
407 downmix_coeffs[1][i] *= norm1;
412 downmix_coeffs[0][i] = (downmix_coeffs[0][i] +
431 int i, j, grp, group_size;
436 group_size = exp_strategy + (exp_strategy ==
EXP_D45);
437 for (grp = 0, i = 0; grp < ngrps; grp++) {
450 for (i = 0, j = 0; i < ngrps * 3; i++) {
451 prevexp += dexp[i] - 2;
456 switch (group_size) {
457 case 4: dexps[j++] = prevexp;
458 dexps[j++] = prevexp;
459 case 2: dexps[j++] = prevexp;
460 case 1: dexps[j++] = prevexp;
477 int band_start = bin;
482 for (bin = band_start; bin < band_end; bin++) {
487 for (bin = band_start; bin < band_end; bin++)
515 int end_freq = s->
end_freq[ch_index];
517 int8_t *exps = s->
dexps[ch_index];
523 for (freq = start_freq; freq < end_freq; freq++) {
524 int bap = baps[freq];
584 coeffs[freq] = mantissa >> exps[freq];
615 if (CONFIG_EAC3_DECODER && !blk)
617 for (bin = s->
start_freq[ch]; bin < s->end_freq[ch]; bin++) {
634 for (ch = 1; ch <= s->
channels; ch++) {
690 for (ch = 1; ch <= channels; ch++) {
694 for (i = 0; i < 128; i++)
698 s->
fdsp->vector_fmul_window_scaled(s->
outptr[ch - 1], s->
delay[ch - 1],
704 for (i = 0; i < 128; i++)
710 s->
fdsp->vector_fmul_window_scaled(s->
outptr[ch - 1], s->
delay[ch - 1],
726 int channel_data_size =
sizeof(s->
delay[0]);
731 memcpy(s->
delay[1], s->
delay[0], channel_data_size);
734 memset(s->
delay[3], 0, channel_data_size);
736 memset(s->
delay[2], 0, channel_data_size);
739 memset(s->
delay[4], 0, channel_data_size);
741 memset(s->
delay[3], 0, channel_data_size);
743 memcpy(s->
delay[2], s->
delay[1], channel_data_size);
744 memset(s->
delay[1], 0, channel_data_size);
767 int ecpl,
int start_subband,
int end_subband,
768 const uint8_t *default_band_struct,
769 int *num_bands,
uint8_t *band_sizes,
770 uint8_t *band_struct,
int band_struct_size)
772 int subbnd, bnd, n_subbands, n_bands=0;
775 n_subbands = end_subband - start_subband;
778 memcpy(band_struct, default_band_struct, band_struct_size);
780 av_assert0(band_struct_size >= start_subband + n_subbands);
782 band_struct += start_subband + 1;
786 for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) {
794 if (num_bands || band_sizes ) {
795 n_bands = n_subbands;
796 bnd_sz[0] = ecpl ? 6 : 12;
797 for (bnd = 0, subbnd = 1; subbnd < n_subbands; subbnd++) {
798 int subbnd_size = (ecpl && subbnd < 4) ? 6 : 12;
799 if (band_struct[subbnd - 1]) {
801 bnd_sz[bnd] += subbnd_size;
803 bnd_sz[++bnd] = subbnd_size;
810 *num_bands = n_bands;
812 memcpy(band_sizes, bnd_sz, n_bands);
819 int dst_start_freq, dst_end_freq, src_start_freq,
820 start_subband, end_subband,
ch;
826 for (ch = 1; ch <= fbw_channels; ch++)
833 start_subband =
get_bits(bc, 3) + 2;
834 if (start_subband > 7)
835 start_subband += start_subband - 7;
841 end_subband += end_subband - 7;
842 dst_start_freq = dst_start_freq * 12 + 25;
843 src_start_freq = start_subband * 12 + 25;
844 dst_end_freq = end_subband * 12 + 25;
847 if (start_subband >= end_subband) {
849 "range (%d >= %d)\n", start_subband, end_subband);
852 if (dst_start_freq >= src_start_freq) {
854 "copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq);
864 start_subband, end_subband,
878 for (ch = 1; ch <= fbw_channels; ch++) {
882 int bin, master_spx_coord;
886 master_spx_coord =
get_bits(bc, 2) * 3;
891 int spx_coord_exp, spx_coord_mant;
895 int64_t accu = ((bin << 23) + (bandsize << 22))
897 nratio = (
int)(accu >> 32);
898 nratio -= spx_blend << 18;
903 }
else if (nratio > 0x7fffff) {
908 accu = (int64_t)nblend * 1859775393;
909 nblend = (
int)((accu + (1<<29)) >> 30);
916 nratio = ((float)((bin + (bandsize >> 1))) / s->
spx_dst_end_freq) - spx_blend;
917 nratio = av_clipf(nratio, 0.0f, 1.0f);
918 nblend = sqrtf(3.0f * nratio);
920 sblend = sqrtf(1.0f - nratio);
927 if (spx_coord_exp == 15) spx_coord_mant <<= 1;
928 else spx_coord_mant += 4;
929 spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
933 accu = (int64_t)nblend * spx_coord_mant;
935 accu = (int64_t)sblend * spx_coord_mant;
938 spx_coord = spx_coord_mant * (1.0f / (1 << 23));
963 int cpl_start_subband, cpl_end_subband;
982 for (ch = 1; ch <= fbw_channels; ch++)
991 cpl_start_subband =
get_bits(bc, 4);
994 if (cpl_start_subband >= cpl_end_subband) {
996 cpl_start_subband, cpl_end_subband);
1009 for (ch = 1; ch <= fbw_channels; ch++) {
1025 int cpl_coords_exist = 0;
1027 for (ch = 1; ch <= fbw_channels; ch++) {
1030 int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
1032 cpl_coords_exist = 1;
1033 master_cpl_coord = 3 *
get_bits(bc, 2);
1037 if (cpl_coord_exp == 15)
1041 s->
cpl_coords[
ch][bnd] >>= (cpl_coord_exp + master_cpl_coord);
1045 "be present in block 0\n");
1070 int i, bnd, seg,
ch, ret;
1071 int different_transforms;
1078 different_transforms = 0;
1080 for (ch = 1; ch <= fbw_channels; ch++) {
1083 different_transforms = 1;
1089 for (ch = 1; ch <= fbw_channels; ch++) {
1102 if (range_bits <= 127 || s->drc_scale <= 1.0)
1106 }
else if (blk == 0) {
1121 for (ch = 1; ch <= fbw_channels; ch++) {
1135 }
else if (!s->
eac3) {
1138 "be present in block 0\n");
1165 "new rematrixing strategy not present in block 0\n");
1171 for (ch = !cpl_in_use; ch <= s->
channels; ch++) {
1175 bit_alloc_stages[
ch] = 3;
1179 for (ch = 1; ch <= fbw_channels; ch++) {
1189 int bandwidth_code =
get_bits(gbc, 6);
1190 if (bandwidth_code > 60) {
1198 if (blk > 0 && s->
end_freq[ch] != prev)
1208 for (ch = !cpl_in_use; ch <= s->
channels; ch++) {
1229 for (ch = !cpl_in_use; ch <= s->
channels; ch++)
1230 bit_alloc_stages[ch] =
FFMAX(bit_alloc_stages[ch], 2);
1233 "be present in block 0\n");
1239 if (!s->
eac3 || !blk) {
1243 csnr = (
get_bits(gbc, 6) - 15) << 4;
1244 for (i = ch = !cpl_in_use; ch <= s->
channels; ch++) {
1247 snr = (csnr +
get_bits(gbc, 4)) << 2;
1250 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[ch], 1);
1260 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[ch], 2);
1263 }
else if (!s->
eac3 && !blk) {
1271 for (ch = !cpl_in_use; ch <= s->
channels; ch++) {
1276 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[ch], 2);
1278 }
else if (s->
eac3 && !blk) {
1279 for (ch = !cpl_in_use; ch <= s->
channels; ch++)
1301 }
else if (!s->
eac3 && !blk) {
1303 "be present in block 0\n");
1312 for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
1318 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[ch], 2);
1321 for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
1330 bit_alloc_stages[
ch] =
FFMAX(bit_alloc_stages[ch], 2);
1333 }
else if (blk == 0) {
1334 for (ch = 0; ch <= s->
channels; ch++) {
1340 for (ch = !cpl_in_use; ch <= s->
channels; ch++) {
1341 if (bit_alloc_stages[ch] > 2) {
1347 if (bit_alloc_stages[ch] > 1) {
1360 if (bit_alloc_stages[ch] > 0) {
1368 bap_tab, s->
bap[ch]);
1389 for (ch = 1; ch <= s->
channels; ch++) {
1390 int audio_channel = 0;
1393 audio_channel = 2-
ch;
1404 gain *= 1.0 / 4194304.0f;
1421 if (different_transforms) {
1431 if (downmix_output) {
1441 if (downmix_output) {
1462 int *got_frame_ptr,
AVPacket *avpkt)
1466 int buf_size = avpkt->
size;
1468 int blk,
ch, err, ret;
1476 if (buf_size >= 2 &&
AV_RB16(buf) == 0x770B) {
1480 (
const uint16_t *) buf, cnt);
1517 "unsupported frame type %d: skipping frame\n",
1608 for (ch = 0; ch < s->
channels; ch++) {
1609 if (ch < s->out_channels)
1621 output[ch] = s->
outptr[channel_map[ch]];
1623 if (!ch || channel_map[ch])
1709 #define OFFSET(x) offsetof(AC3DecodeContext, x)
1710 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
SHORTFLOAT * downmix_coeffs[2]
stereo downmix coefficients
static int coupling_strategy(AC3DecodeContext *s, int blk, uint8_t *bit_alloc_stages)
void ff_ac3dsp_downmix(AC3DSPContext *c, float **samples, float **matrix, int out_ch, int in_ch, int len)
const uint8_t ff_ac3_bap_tab[64]
static const uint8_t ac3_default_coeffs[8][5][2]
Table for default stereo downmixing coefficients reference: Section 7.8.2 Downmixing Into Two Channel...
int dba_nsegs[AC3_MAX_CHANNELS]
number of delta segments
uint8_t input_buffer[AC3_FRAME_BUFFER_SIZE+AV_INPUT_BUFFER_PADDING_SIZE]
temp buffer to prevent overread
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
int16_t psd[AC3_MAX_CHANNELS][AC3_MAX_COEFS]
scaled exponents
int spx_in_use
spectral extension in use (spxinu)
ptrdiff_t const GLvoid * data
static int spx_strategy(AC3DecodeContext *s, int blk)
uint8_t dba_offsets[AC3_MAX_CHANNELS][8]
delta segment offsets
const uint8_t ff_eac3_default_spx_band_struct[17]
Table E2.15 Default Spectral Extension Banding Structure.
const uint8_t ff_ac3_slow_decay_tab[4]
int dither_flag[AC3_MAX_CHANNELS]
dither flags (dithflg)
int16_t mask[AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS]
masking curve values
int preferred_downmix
Preferred 2-channel downmix mode (dmixmod)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
int64_t bit_rate
the average bitrate
const uint8_t ff_ac3_ungroup_3_in_5_bits_tab[32][3]
Table used to ungroup 3 values stored in 5 bits.
static void skip_bits_long(GetBitContext *s, int n)
void(* bswap16_buf)(uint16_t *dst, const uint16_t *src, int len)
double center_mix_level_ltrt
Absolute scale factor representing the nominal level of the center channel during an Lt/Rt compatible...
static const uint8_t bap_tab[64]
#define LEVEL_PLUS_1POINT5DB
AVDownmixInfo * av_downmix_info_update_side_data(AVFrame *frame)
Get a frame's AV_FRAME_DATA_DOWNMIX_INFO side data for editing.
static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m)
Decode the transform coefficients for a particular channel reference: Section 7.3 Quantization and De...
int channels
number of total channels
int spx_dst_end_freq
spx end frequency bin
static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch, mant_groups *m)
int exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS]
exponent strategies (expstr)
int lfe_on
lfe channel in use
#define AV_CH_LAYOUT_STEREO
const uint16_t ff_ac3_slow_gain_tab[4]
int block_switch[AC3_MAX_CHANNELS]
block switch flags (blksw)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
int dba_syntax
delta bit allocation syntax enabled (dbaflde)
int dialog_normalization[2]
dialog level in dBFS (dialnorm)
int spx_src_start_freq
spx start frequency bin
int heavy_compression
apply heavy compression
static int get_sbits(GetBitContext *s, int n)
static void spx_coordinates(AC3DecodeContext *s)
uint8_t cpl_band_sizes[AC3_MAX_CPL_BANDS]
number of coeffs in each coupling band
static void scale_coefs(int32_t *dst, const int32_t *src, int dynrng, int len)
SHORTFLOAT output[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]
output after imdct transform and windowing
#define AC3_FRAME_BUFFER_SIZE
Large enough for maximum possible frame size when the specification limit is ignored.
enum AVAudioServiceType audio_service_type
Type of service that the audio stream conveys.
int lfe_mix_level_exists
indicates if lfemixlevcod is specified (lfemixlevcode)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
uint8_t spx_band_struct[SPX_MAX_BANDS]
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
enum AVSampleFormat sample_fmt
audio sample format
float ltrt_center_mix_level
int first_cpl_coords[AC3_MAX_CHANNELS]
first coupling coordinates states (firstcplcos)
static av_cold int end(AVCodecContext *avctx)
INTFLOAT spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]
spx noise blending factor (nblendfact)
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
static void do_imdct(AC3DecodeContext *s, int channels)
Inverse MDCT Transform.
static av_cold int ac3_decode_end(AVCodecContext *avctx)
Uninitialize the AC-3 decoder.
static uint8_t ungroup_3_in_7_bits_tab[128][3]
table for ungrouping 3 values in 7 bits.
static void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch)
Decode mantissas in a single channel for the entire frame.
int num_rematrixing_bands
number of rematrixing bands (nrematbnd)
#define AV_CH_LOW_FREQUENCY
int fast_gain[AC3_MAX_CHANNELS]
fast gain values/SMR's (fgain)
Public header for CRC hash function implementation.
float ltrt_surround_mix_level
double surround_mix_level_ltrt
Absolute scale factor representing the nominal level of the surround channels during an Lt/Rt compati...
int cpl_coords[AC3_MAX_CHANNELS][AC3_MAX_CPL_BANDS]
coupling coordinates (cplco)
static int b1_mantissas[32][3]
tables for ungrouping mantissas
int num_exp_groups[AC3_MAX_CHANNELS]
Number of exponent groups (nexpgrp)
Lt/Rt 2-channel downmix, Dolby Pro Logic II compatible.
#define LEVEL_MINUS_1POINT5DB
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
static int ff_eac3_parse_header(AC3DecodeContext *s)
Parse the E-AC-3 frame header.
int bit_allocation_syntax
bit allocation model syntax enabled (bamode)
int phase_flags_in_use
phase flags in use (phsflginu)
double lfe_mix_level
Absolute scale factor representing the level at which the LFE data is mixed into L/R channels during ...
Common code between the AC-3 and E-AC-3 decoders.
Grouped mantissas for 3-level 5-level and 11-level quantization.
INTFLOAT delay[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]
delay - added to the next block
int phase_flags[AC3_MAX_CPL_BANDS]
phase flags (phsflg)
int out_channels
number of output channels
const uint16_t avpriv_ac3_channel_layout_tab[8]
Map audio coding mode (acmod) to channel layout mask.
This structure describes optional metadata relevant to a downmix procedure.
int consistent_noise_generation
seed noise generation with AC-3 frame on decode
int substreamid
substream identification
FFTContext imdct_256
for 256 sample IMDCT
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int decode_audio_block(AC3DecodeContext *s, int blk)
Decode a single audio block from the AC-3 bitstream.
#define AC3_DYNAMIC_RANGE1
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
const uint8_t ff_ac3_fast_decay_tab[4]
int bit_rate
stream bit rate, in bits-per-second
int lfe_ch
index of LFE channel
uint8_t first_spx_coords[AC3_MAX_CHANNELS]
first spx coordinates states (firstspxcos)
AC3BitAllocParameters bit_alloc_params
bit allocation parameters
int dolby_surround_mode
dolby surround mode (dsurmod)
static void decode_transform_coeffs(AC3DecodeContext *s, int blk)
Decode the transform coefficients.
GetBitContext gbc
bitstream reader
static const uint8_t dither[8][8]
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
#define AC3_MAX_CHANNELS
maximum number of channels, including coupling channel
int flags
AV_CODEC_FLAG_*.
int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd, int start, int end, int fast_gain, int is_lfe, int dba_mode, int dba_nsegs, uint8_t *dba_offsets, uint8_t *dba_lengths, uint8_t *dba_values, int16_t *mask)
Calculate the masking curve.
int dolby_headphone_mode
dolby headphone mode (dheadphonmod)
float loro_center_mix_level
int compression_exists[2]
compression field is valid for frame (compre)
INTFLOAT transform_coeffs[AC3_MAX_CHANNELS][AC3_MAX_COEFS]
transform coefficients
static int ac3_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Decode a single AC-3 frame.
int block_switch_syntax
block switch syntax enabled (blkswe)
int fast_gain_syntax
fast gain codes enabled (frmfgaincode)
int16_t band_psd[AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS]
interpolated exponents
uint8_t channel_uses_spx[AC3_MAX_CHANNELS]
channel uses spectral extension (chinspx)
uint64_t channel_layout
Audio channel layout.
#define LEVEL_MINUS_4POINT5DB
static int b3_mantissas[8]
static int set_downmix_coeffs(AC3DecodeContext *s)
Set stereo downmixing coefficients based on frame header info.
INTFLOAT dynamic_range[2]
dynamic range
static float dynamic_range_tab[256]
dynamic range table.
INTFLOAT window[AC3_BLOCK_SIZE]
window coefficients
static void do_rematrixing(AC3DecodeContext *s)
Stereo rematrixing.
#define AC3_HEAVY_RANGE(x)
SHORTFLOAT * outptr[AC3_MAX_CHANNELS]
audio channel layout utility functions
int surround_mix_level_ltrt
Surround mix level index for Lt/Rt (ltrtsurmixlev)
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
static int b5_mantissas[16]
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
float loro_surround_mix_level
Lt/Rt 2-channel downmix, Dolby Surround compatible.
int dither_flag_syntax
dither flag syntax enabled (dithflage)
int end_freq[AC3_MAX_CHANNELS]
end frequency bin (endmant)
INTFLOAT heavy_dynamic_range[2]
heavy dynamic range compression
av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
INTFLOAT tmp_output[AC3_BLOCK_SIZE]
temporary storage for output before windowing
uint8_t dba_values[AC3_MAX_CHANNELS][8]
delta values for each segment
FFTContext imdct_512
for 512 sample IMDCT
const uint16_t ff_ac3_fast_gain_tab[8]
uint32_t av_crc(const AVCRC *ctx, uint32_t crc, const uint8_t *buffer, size_t length)
Calculate the CRC of a block.
void(* int32_to_float_fmul_scalar)(float *dst, const int32_t *src, float mul, int len)
Convert an array of int32_t to float and multiply by a float value.
static int b2_mantissas[128][3]
Lo/Ro 2-channel downmix (Stereo).
#define AV_EF_EXPLODE
abort decoding on minor error detection
av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
AVCodecContext * avctx
parent context
static int b4_mantissas[128][2]
static void decode_band_structure(GetBitContext *gbc, int blk, int eac3, int ecpl, int start_subband, int end_subband, const uint8_t *default_band_struct, int *num_bands, uint8_t *band_sizes, uint8_t *band_struct, int band_struct_size)
Decode band structure for coupling, spectral extension, or enhanced coupling.
int ff_side_data_update_matrix_encoding(AVFrame *frame, enum AVMatrixEncoding matrix_encoding)
Add or update AV_FRAME_DATA_MATRIXENCODING side data.
int av_lfg_init_from_data(AVLFG *c, const uint8_t *data, unsigned int length)
Seed the state of the ALFG using binary data.
int channel_in_cpl[AC3_MAX_CHANNELS]
channel in coupling (chincpl)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
int fbw_channels
number of full-bandwidth channels
void(* bit_alloc_calc_bap)(int16_t *mask, int16_t *psd, int start, int end, int snr_offset, int floor, const uint8_t *bap_tab, uint8_t *bap)
Calculate bit allocation pointers.
INTFLOAT * xcfptr[AC3_MAX_CHANNELS]
uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS]
bit allocation pointers
double surround_mix_level
Absolute scale factor representing the nominal level of the surround channels during a regular downmi...
#define CPL_CH
coupling channel index
const uint8_t ff_eac3_default_cpl_band_struct[18]
Table E2.16 Default Coupling Banding Structure.
uint8_t cpl_band_struct[AC3_MAX_CPL_BANDS]
int sample_rate
samples per second
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
int num_cpl_bands
number of coupling bands (ncplbnd)
int num_spx_bands
number of spx bands (nspxbnds)
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
static const float gain_levels[9]
Adjustments in dB gain.
#define AV_EF_CAREFUL
consider things that violate the spec, are fast to calculate and have not been seen in the wild as er...
static void ff_eac3_apply_spectral_extension(AC3DecodeContext *s)
Apply spectral extension to each channel by copying lower frequency coefficients to higher frequency ...
static unsigned int get_bits1(GetBitContext *s)
static const int end_freq_inv_tab[8]
INTFLOAT spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]
spx signal blending factor (sblendfact)
static void skip_bits(GetBitContext *s, int n)
int bitstream_mode
bitstream mode (bsmod)
int center_mix_level_ltrt
Center mix level index for Lt/Rt (ltrtcmixlev)
int frame_type
frame type (strmtyp)
static int decode_exponents(AC3DecodeContext *s, GetBitContext *gbc, int exp_strategy, int ngrps, uint8_t absexp, int8_t *dexps)
Decode the grouped exponents according to exponent strategy.
static const uint8_t quantization_tab[16]
Quantization table: levels for symmetric.
#define AV_EF_CRCCHECK
Verify checksums embedded in the bitstream (could be of either encoded or decoded data...
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
#define FF_DECODE_ERROR_INVALID_BITSTREAM
int eac3
indicates if current frame is E-AC-3
int channel_uses_aht[AC3_MAX_CHANNELS]
channel AHT in use (chahtinu)
const int16_t ff_ac3_floor_tab[8]
int decode_error_flags
decode error flags of the frame, set to a combination of FF_DECODE_ERROR_xxx flags if the decoder pro...
uint8_t dba_lengths[AC3_MAX_CHANNELS][8]
delta segment lengths
static int coupling_coordinates(AC3DecodeContext *s, int blk)
static int ac3_parse_header(AC3DecodeContext *s)
Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
int cpl_in_use[AC3_MAX_BLOCKS]
coupling in use (cplinu)
#define AC3_DYNAMIC_RANGE(x)
int dba_mode[AC3_MAX_CHANNELS]
delta bit allocation mode
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
int first_cpl_leak
first coupling leak state (firstcplleak)
int surround_mix_level
Surround mix level index.
int snr_offset[AC3_MAX_CHANNELS]
signal-to-noise ratio offsets (snroffst)
int downmixed
indicates if coeffs are currently downmixed
common internal api header.
INTFLOAT * dlyptr[AC3_MAX_CHANNELS]
int num_blocks
number of audio blocks
FmtConvertContext fmt_conv
optimized conversion functions
static void ac3_downmix_c_fixed16(int16_t **samples, int16_t **matrix, int out_ch, int in_ch, int len)
Downmix samples from original signal to stereo or mono (this is for 16-bit samples and fixed point de...
static av_cold void ac3_tables_init(void)
int target_level
target level in dBFS
double center_mix_level
Absolute scale factor representing the nominal level of the center channel during a regular downmix...
int pre_mantissa[AC3_MAX_CHANNELS][AC3_MAX_COEFS][AC3_MAX_BLOCKS]
pre-IDCT mantissas
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd, int16_t *band_psd)
Calculate the log power-spectral density of the input signal.
int rematrixing_flags[4]
rematrixing flags (rematflg)
int fixed_coeffs[AC3_MAX_CHANNELS][AC3_MAX_COEFS]
fixed-point transform coefficients
uint8_t spx_band_sizes[SPX_MAX_BANDS]
number of bins in each spx band
int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo **phdr)
Parse AC-3 frame header.
int lfe_mix_level
LFE mix level index (lfemixlevcod)
static int parse_frame_header(AC3DecodeContext *s)
Common function to parse AC-3 or E-AC-3 frame header.
int snr_offset_strategy
SNR offset strategy (snroffststr)
static const int16_t coeffs[]
int start_freq[AC3_MAX_CHANNELS]
start frequency bin (strtmant)
int channels
number of audio channels
int sample_rate
sample frequency, in Hz
int center_mix_level
Center mix level index.
int output_mode
output channel configuration
static void calc_transform_coeffs_cpl(AC3DecodeContext *s)
Generate transform coefficients for each coupled channel in the coupling range using the coupling coe...
static const float gain_levels_lfe[32]
Adjustments in dB gain (LFE, +10 to -21 dB)
int frame_size
current frame size, in bytes
static int symmetric_dequant(int code, int levels)
Symmetrical Dequantization reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantizati...
static void ac3_upmix_delay(AC3DecodeContext *s)
Upmix delay samples from stereo to original channel layout.
const uint8_t ff_ac3_rematrix_band_tab[5]
Table of bin locations for rematrixing bands reference: Section 7.5.2 Rematrixing : Frequency Band De...
const uint8_t ff_eac3_hebap_tab[64]
int dolby_surround_ex_mode
dolby surround ex mode (dsurexmod)
#define av_malloc_array(a, b)
const uint8_t ff_ac3_dec_channel_map[8][2][6]
Table to remap channels from AC-3 order to SMPTE order.
AVLFG dith_state
for dither generation
int cpl_strategy_exists[AC3_MAX_BLOCKS]
coupling strategy exists (cplstre)
static av_cold int ac3_decode_init(AVCodecContext *avctx)
AVCodec initialization.
const uint16_t ff_ac3_db_per_bit_tab[4]
int bitstream_id
bitstream id (bsid)
#define AV_CH_LAYOUT_MONO
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
static void remove_dithering(AC3DecodeContext *s)
Remove random dithering from coupling range coefficients with zero-bit mantissas for coupled channels...
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
int channel_mode
channel mode (acmod)
float ff_ac3_heavy_dynamic_range_tab[256]
int spx_dst_start_freq
spx starting frequency bin for copying (copystartmant) the copy region ends at the start of the spx r...
int skip_syntax
skip field syntax enabled (skipflde)
enum AVDownmixType preferred_downmix_type
Type of downmix preferred by the mastering engineer.
int8_t dexps[AC3_MAX_CHANNELS][AC3_MAX_COEFS]
decoded exponents