50 #define IMC_BLOCK_SIZE 64 51 #define IMC_FRAME_ID 0x21 106 float weights1[31], weights2[31];
113 #define VLC_TABLES_SIZE 9512 116 0, 640, 1156, 1732, 2308, 2852, 3396, 3924,
124 return 3.5 * atan((freq / 7500.0) * (freq / 7500.0)) + 13.0 * atan(freq * 0.00076);
129 double freqmin[32], freqmid[32], freqmax[32];
130 double scale = sampling_rate / (256.0 * 2.0 * 2.0);
131 double nyquist_freq = sampling_rate * 0.5;
132 double freq, bark, prev_bark = 0,
tf,
tb;
135 for (i = 0; i < 32; i++) {
140 tb = bark - prev_bark;
149 while (
tf < nyquist_freq) {
161 if (tb <= bark - 0.5)
167 for (i = 0; i < 32; i++) {
169 for (j = 31; j > 0 && freq <= freqmid[j]; j--);
173 for (j = 0; j < 32 && freq >= freqmid[j]; j++);
186 "Strange sample rate of %i, file likely corrupt or " 187 "needing a new table derivation method.\n",
200 for (j = 0; j < avctx->
channels; j++) {
203 for (i = 0; i <
BANDS; i++)
212 for (i = 0; i <
COEFFS; i++)
214 for (i = 0; i < COEFFS / 2; i++) {
218 r1 = sin((i * 4.0 + 1.0) / 1024.0 *
M_PI);
219 r2 = cos((i * 4.0 + 1.0) / 1024.0 *
M_PI);
232 for (i = 0; i < 30; i++)
236 for (i = 0; i < 4 ; i++) {
237 for (j = 0; j < 4; j++) {
239 huffman_vlc[
i][j].
table_allocated = vlc_offsets[i * 4 + j + 1] - vlc_offsets[i * 4 + j];
281 float snr_limit = 1.e-30;
285 for (i = 0; i <
BANDS; i++) {
286 flcoeffs5[
i] = workT2[
i] = 0.0;
288 workT1[
i] = flcoeffs1[
i] * flcoeffs1[
i];
289 flcoeffs3[
i] = 2.0 * flcoeffs2[
i];
292 flcoeffs3[
i] = -30000.0;
294 workT3[
i] = bandWidthT[
i] * workT1[
i] * 0.01;
295 if (workT3[i] <= snr_limit)
299 for (i = 0; i <
BANDS; i++) {
300 for (cnt2 = i; cnt2 < q->
cyclTab[
i]; cnt2++)
301 flcoeffs5[cnt2] = flcoeffs5[cnt2] + workT3[i];
302 workT2[cnt2 - 1] = workT2[cnt2 - 1] + workT3[
i];
305 for (i = 1; i <
BANDS; i++) {
306 accum = (workT2[i - 1] + accum) * q->
weights1[i - 1];
307 flcoeffs5[i] += accum;
310 for (i = 0; i <
BANDS; i++)
313 for (i = 0; i <
BANDS; i++) {
314 for (cnt2 = i - 1; cnt2 > q->
cyclTab2[
i]; cnt2--)
315 flcoeffs5[cnt2] += workT3[i];
316 workT2[cnt2+1] += workT3[
i];
321 for (i = BANDS-2; i >= 0; i--) {
322 accum = (workT2[i+1] + accum) * q->
weights2[i];
323 flcoeffs5[i] += accum;
338 s = stream_format_code >> 1;
339 hufftab[0] = &huffman_vlc[
s][0];
340 hufftab[1] = &huffman_vlc[
s][1];
341 hufftab[2] = &huffman_vlc[
s][2];
342 hufftab[3] = &huffman_vlc[
s][3];
345 if (stream_format_code & 4)
349 for (i = start; i <
BANDS; i++) {
351 hufftab[cb_sel[i]]->
bits, 2);
352 if (levlCoeffs[i] == 17)
364 for (i = 1; i <
BANDS; i++)
375 flcoeffs1[0] = 20000.0 /
exp2 (levlCoeffBuf[0] * 0.18945);
376 flcoeffs2[0] =
log2f(flcoeffs1[0]);
380 for (i = 1; i <
BANDS; i++) {
381 level = levlCoeffBuf[
i];
388 else if (level <= 24)
394 tmp2 += 0.83048 *
level;
411 for (i = 0; i <
BANDS; i++) {
413 if (levlCoeffBuf[i] < 16) {
415 flcoeffs2[
i] = (levlCoeffBuf[
i] - 7) * 0.83048 + flcoeffs2[i];
417 flcoeffs1[
i] = old_floor[
i];
429 flcoeffs1[
pos] = 20000.0 / pow (2, levlCoeffBuf[0] * 0.18945);
430 flcoeffs2[
pos] =
log2f(flcoeffs1[pos]);
431 tmp = flcoeffs1[
pos];
432 tmp2 = flcoeffs2[
pos];
435 for (i = 0; i <
BANDS; i++) {
438 level = *levlCoeffBuf++;
439 flcoeffs1[
i] = tmp *
powf(10.0, -level * 0.4375);
440 flcoeffs2[
i] = tmp2 - 1.4533435415 *
level;
448 int stream_format_code,
int freebits,
int flag)
451 const float limit = -1.e20;
460 float lowest = 1.e10;
466 for (i = 0; i <
BANDS; i++)
469 for (i = 0; i < BANDS - 1; i++) {
478 highest = highest * 0.25;
480 for (i = 0; i <
BANDS; i++) {
497 if (stream_format_code & 0x2) {
504 for (i = (stream_format_code & 0x2) ? 4 : 0; i < BANDS - 1; i++) {
513 summa = (summa * 0.5 - freebits) / iacc;
516 for (i = 0; i < BANDS / 2; i++) {
517 rres = summer - freebits;
518 if ((rres >= -8) && (rres <= 8))
524 for (j = (stream_format_code & 0x2) ? 4 : 0; j <
BANDS; j++) {
525 cwlen = av_clipf(((chctx->
flcoeffs4[j] * 0.5) - summa + 0.5), 0, 6);
536 if (freebits < summer)
543 summa = (float)(summer - freebits) / ((t1 + 1) * iacc) + summa;
546 for (i = (stream_format_code & 0x2) ? 4 : 0; i <
BANDS; i++) {
551 if (freebits > summer) {
552 for (i = 0; i <
BANDS; i++) {
560 if (highest <= -1.e20)
566 for (i = 0; i <
BANDS; i++) {
567 if (workT[i] > highest) {
573 if (highest > -1.e20) {
574 workT[found_indx] -= 2.0;
576 workT[found_indx] = -1.e20;
578 for (j =
band_tab[found_indx]; j < band_tab[found_indx + 1] && (freebits > summer); j++) {
583 }
while (freebits > summer);
585 if (freebits < summer) {
586 for (i = 0; i <
BANDS; i++) {
590 if (stream_format_code & 0x2) {
596 while (freebits < summer) {
599 for (i = 0; i <
BANDS; i++) {
600 if (workT[i] < lowest) {
607 workT[low_indx] = lowest + 2.0;
610 workT[low_indx] = 1.e20;
612 for (j =
band_tab[low_indx]; j <
band_tab[low_indx+1] && (freebits < summer); j++) {
629 for (i = 0; i <
BANDS; i++) {
636 for (j = band_tab[i]; j < band_tab[i + 1]; j++) {
667 if (j < band_tab[i + 1]) {
688 for (i = 0; i <
BANDS; i++) {
693 while (corrected < summer) {
694 if (highest <= -1.e20)
699 for (i = 0; i <
BANDS; i++) {
700 if (workT[i] > highest) {
706 if (highest > -1.e20) {
707 workT[found_indx] -= 2.0;
708 if (++(chctx->
bitsBandT[found_indx]) == 6)
709 workT[found_indx] = -1.e20;
711 for (j =
band_tab[found_indx]; j <
band_tab[found_indx+1] && (corrected < summer); j++) {
729 for (i = 0; i <
COEFFS / 2; i++) {
741 for (i = 0; i <
COEFFS / 2; i++) {
755 int stream_format_code)
758 int middle_value, cw_len, max_size;
759 const float *quantizer;
761 for (i = 0; i <
BANDS; i++) {
769 max_size = 1 << cw_len;
770 middle_value = max_size >> 1;
797 int i, j, cw_len, cw;
799 for (i = 0; i <
BANDS; i++) {
810 "Potential problem on band %i, coefficient %i" 811 ": cw_len=%i\n", i, j, cw_len);
827 for (i = 0; i <
BANDS; i++) {
833 if (((
int)((band_tab[i + 1] - band_tab[i]) * 1.5) > chctx->
sumLenArr[i]) && (chctx->
sumLenArr[i] > 0))
839 for (i = 0; i <
BANDS; i++) {
851 for (i = 0; i <
BANDS; i++) {
868 int stream_format_code;
869 int imc_hdr,
i, j, ret;
872 int counter, bitscount;
878 if (imc_hdr & 0x18) {
885 if (stream_format_code & 0x04)
889 for (i = 0; i <
BANDS; i++)
897 if (stream_format_code & 0x1)
902 if (stream_format_code & 0x1)
905 else if (stream_format_code & 0x4)
912 for(i=0; i<
BANDS; i++) {
922 if (stream_format_code & 0x1) {
923 for (i = 0; i <
BANDS; i++) {
930 for (i = 0; i <
BANDS; i++) {
939 for (i = 0; i < BANDS - 1; i++)
950 if (stream_format_code & 0x2) {
957 for (i = 1; i < 4; i++) {
958 if (stream_format_code & 0x1)
971 if (!(stream_format_code & 0x2))
983 if (stream_format_code & 0x1) {
984 for (i = 0; i <
BANDS; i++)
990 for (i = 0; i <
BANDS; i++) {
1016 int *got_frame_ptr,
AVPacket *avpkt)
1020 int buf_size = avpkt->
size;
1029 if (buf_size < IMC_BLOCK_SIZE * avctx->
channels) {
1039 for (i = 0; i < avctx->
channels; i++) {
1080 #if CONFIG_IMC_DECODER 1096 #if CONFIG_IAC_DECODER int skipFlags[COEFFS]
skip coefficient decoding or not
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
int codewords[COEFFS]
raw codewords read from bitstream
void(* bswap16_buf)(uint16_t *dst, const uint16_t *src, int len)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
int skipFlagRaw[BANDS]
skip flags are stored in raw form or not
static av_cold int init(AVCodecContext *avctx)
static const int vlc_offsets[17]
#define avpriv_request_sample(...)
float mdct_sine_window[COEFFS]
MDCT tables.
static const uint8_t imc_huffman_lens[4][4][18]
int skipFlagCount[BANDS]
skipped coefficients per band
static const float imc_weights2[31]
#define AV_CH_LAYOUT_STEREO
static void imc_read_level_coeffs(IMCContext *q, int stream_format_code, int *levlCoeffs)
#define init_vlc(vlc, nb_bits, nb_codes, bits, bits_wrap, bits_size, codes, codes_wrap, codes_size, flags)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
enum AVSampleFormat sample_fmt
audio sample format
int bandFlagsBuf[BANDS]
flags for each band
static av_cold int imc_decode_close(AVCodecContext *avctx)
static const int8_t cyclTab[32]
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
static int get_bits_count(const GetBitContext *s)
static const float imc_weights1[31]
bitstream reader API header.
static void imc_get_skip_coeff(IMCContext *q, IMCChannel *chctx)
static void imc_refine_bit_allocation(IMCContext *q, IMCChannel *chctx)
#define i(width, name, range_min, range_max)
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void(* fft_permute)(struct FFTContext *s, FFTComplex *z)
Do the permutation needed BEFORE calling fft_calc().
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
static const float *const imc_exp_tab2
static void imc_decode_level_coefficients_raw(IMCContext *q, int *levlCoeffBuf, float *flcoeffs1, float *flcoeffs2)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static void imc_imdct256(IMCContext *q, IMCChannel *chctx, int channels)
int flags
AV_CODEC_FLAG_*.
static void imc_calculate_coeffs(IMCContext *q, float *flcoeffs1, float *flcoeffs2, int *bandWidthT, float *flcoeffs3, float *flcoeffs5)
const char * name
Name of the codec implementation.
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { 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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) 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);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=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) { int planes=out->planar ? out->ch_count :1;unsigned m=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){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
uint64_t channel_layout
Audio channel layout.
static const int8_t cyclTab2[32]
static int imc_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static void imc_decode_level_coefficients2(IMCContext *q, int *levlCoeffBuf, float *old_floor, float *flcoeffs1, float *flcoeffs2)
common internal API header
audio channel layout utility functions
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
static const uint16_t band_tab[33]
int bitsBandT[BANDS]
how many bits per codeword in band
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
float last_fft_im[COEFFS]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static void imc_get_coeffs(AVCodecContext *avctx, IMCContext *q, IMCChannel *chctx)
#define AV_LOG_INFO
Standard information.
static const float xTab[14]
FFTComplex samples[COEFFS/2]
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
static int inverse_quant_coeff(IMCContext *q, IMCChannel *chctx, int stream_format_code)
int sample_rate
samples per second
void AAC_RENAME() ff_sine_window_init(INTFLOAT *window, int n)
Generate a sine window.
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
static double freq2bark(double freq)
static unsigned int get_bits1(GetBitContext *s)
int bandWidthT[BANDS]
codewords per band
static av_cold void flush(AVCodecContext *avctx)
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static VLC huffman_vlc[4][4]
static VLC_TYPE vlc_tables[VLC_TABLES_SIZE][2]
static const float imc_quantizer1[4][8]
static av_cold void iac_generate_tabs(IMCContext *q, int sampling_rate)
internal math functions header
common internal api header.
static int imc_decode_block(AVCodecContext *avctx, IMCContext *q, int ch)
#define INIT_VLC_USE_NEW_STATIC
void(* fft_calc)(struct FFTContext *s, FFTComplex *z)
Do a complex FFT with the parameters defined in ff_fft_init().
static int bit_allocation(IMCContext *q, IMCChannel *chctx, int stream_format_code, int freebits, int flag)
Perform bit allocation depending on bits available.
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
static const uint8_t imc_cb_select[4][32]
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
static enum AVSampleFormat sample_fmts[]
#define LOCAL_ALIGNED_16(t, v,...)
static void imc_adjust_bit_allocation(IMCContext *q, IMCChannel *chctx, int summer)
Increase highest' band coefficient sizes as some bits won't be used.
static const float imc_quantizer2[2][56]
int sumLenArr[BANDS]
bits for all coeffs in band
static const uint8_t imc_huffman_sizes[4]
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
static void imc_read_level_coeffs_raw(IMCContext *q, int stream_format_code, int *levlCoeffs)
This structure stores compressed data.
static void imc_decode_level_coefficients(IMCContext *q, int *levlCoeffBuf, float *flcoeffs1, float *flcoeffs2)
int skipFlagBits[BANDS]
bits used to code skip flags
static av_cold int imc_decode_init(AVCodecContext *avctx)
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
static const float imc_exp_tab[32]
static const uint16_t imc_huffman_bits[4][4][18]
int CWlengthT[COEFFS]
how many bits in each codeword