36 #define ON2AVC_SUBFRAME_SIZE 1024 92 int w,
b, band_off = 0;
113 int bits_per_sect = c->
is_long ? 5 : 3;
114 int esc_val = (1 << bits_per_sect) - 1;
118 while (band < num_bands) {
123 if (run > num_bands - band - run_len) {
128 }
while (run == esc_val);
143 int w, w2,
b, scale,
first = 1;
176 if (scale < 0 || scale > 127) {
190 return v * sqrtf(
abs(v)) * scale;
195 int dst_size,
int type,
float band_scale)
199 for (i = 0; i < dst_size; i += 4) {
202 for (j = 0; j < 4; j++) {
203 val1 =
sign_extend((val >> (12 - j * 4)) & 0xF, 4);
228 int dst_size,
int type,
float band_scale)
230 int i,
val, val1, val2, sign;
232 for (i = 0; i < dst_size; i += 2) {
238 if (val1 <= -16 || val1 >= 16) {
239 sign = 1 - (val1 < 0) * 2;
242 if (val2 <= -16 || val2 >= 16) {
243 sign = 1 - (val2 < 0) * 2;
266 coeff_ptr = c->
coeffs[ch];
275 coeff_ptr += band_size;
284 coeff_ptr += band_size;
296 float *ch0 = c->
coeffs[0];
297 float *ch1 = c->
coeffs[1];
301 if (c->
ms_info[band_off + b]) {
303 float l = *ch0,
r = *ch1;
319 memset(src, 0,
sizeof(*src) * order0);
320 memset(src + len - order1, 0,
sizeof(*src) * order1);
324 int step,
int order0,
int order1,
const double *
const *
tabs)
332 for (i = 0; i < tab_step; i++) {
334 for (j = 0; j < order0; j++)
335 sum += src[j] * tab[j * tab_step + i];
339 out = dst + dst_len - tab_step;
341 src2 = src + (dst_len - tab_step) / step + 1 + order0;
342 for (i = 0; i < tab_step; i++) {
344 for (j = 0; j < order1; j++)
345 sum += src2[j] * tab[j * tab_step + i];
351 const double *
tab,
int tab_len,
int step,
352 int order0,
int order1,
const double *
const *
tabs)
358 steps = (src2_len - tab_len) / step + 1;
359 pretwiddle(src1, src2, src2_len, tab_len, step, order0, order1, tabs);
362 for (i = 0; i < steps; i++) {
363 float in0 = src1[order0 +
i];
364 int pos = (src2_len - 1) & mask;
367 const double *t =
tab;
368 for (j = pos; j >= 0; j--)
369 src2[j] += in0 * *t++;
370 for (j = 0; j < tab_len - pos - 1; j++)
371 src2[src2_len - j - 1] += in0 * tab[pos + 1 + j];
373 for (j = 0; j < tab_len; j++)
374 src2[pos - j] += in0 * tab[j];
380 #define CMUL1_R(s, t, is, it) \ 381 s[is + 0] * t[it + 0] - s[is + 1] * t[it + 1] 382 #define CMUL1_I(s, t, is, it) \ 383 s[is + 0] * t[it + 1] + s[is + 1] * t[it + 0] 384 #define CMUL2_R(s, t, is, it) \ 385 s[is + 0] * t[it + 0] + s[is + 1] * t[it + 1] 386 #define CMUL2_I(s, t, is, it) \ 387 s[is + 0] * t[it + 1] - s[is + 1] * t[it + 0] 389 #define CMUL0(dst, id, s0, s1, s2, s3, t0, t1, t2, t3, is, it) \ 390 dst[id] = s0[is] * t0[it] + s1[is] * t1[it] \ 391 + s2[is] * t2[it] + s3[is] * t3[it]; \ 392 dst[id + 1] = s0[is] * t0[it + 1] + s1[is] * t1[it + 1] \ 393 + s2[is] * t2[it + 1] + s3[is] * t3[it + 1]; 395 #define CMUL1(dst, s0, s1, s2, s3, t0, t1, t2, t3, is, it) \ 396 *dst++ = CMUL1_R(s0, t0, is, it) \ 397 + CMUL1_R(s1, t1, is, it) \ 398 + CMUL1_R(s2, t2, is, it) \ 399 + CMUL1_R(s3, t3, is, it); \ 400 *dst++ = CMUL1_I(s0, t0, is, it) \ 401 + CMUL1_I(s1, t1, is, it) \ 402 + CMUL1_I(s2, t2, is, it) \ 403 + CMUL1_I(s3, t3, is, it); 405 #define CMUL2(dst, s0, s1, s2, s3, t0, t1, t2, t3, is, it) \ 406 *dst++ = CMUL2_R(s0, t0, is, it) \ 407 + CMUL2_R(s1, t1, is, it) \ 408 + CMUL2_R(s2, t2, is, it) \ 409 + CMUL2_R(s3, t3, is, it); \ 410 *dst++ = CMUL2_I(s0, t0, is, it) \ 411 + CMUL2_I(s1, t1, is, it) \ 412 + CMUL2_I(s2, t2, is, it) \ 413 + CMUL2_I(s3, t3, is, it); 416 const float *
t0,
const float *
t1,
417 const float *
t2,
const float *
t3,
int len,
int step)
419 const float *h0, *h1, *h2, *h3;
422 int len2 = len >> 1, len4 = len >> 2;
427 for (half = len2; tmp > 1; half <<= 1, tmp >>= 1);
434 CMUL0(dst, 0, s0, s1, s2, s3, t0, t1, t2, t3, 0, 0);
436 hoff = 2 * step * (len4 >> 1);
441 d2 = dst + 2 + (len >> 1);
442 for (i = 0; i < (len4 - 1) >> 1; i++) {
443 CMUL1(d1, s0, s1, s2, s3, t0, t1, t2, t3, j, k);
444 CMUL1(d2, s0, s1, s2, s3, h0, h1, h2, h3, j, k);
448 CMUL0(dst, len4, s0, s1, s2, s3, t0, t1, t2, t3, 1, hoff);
449 CMUL0(dst, len4 + len2, s0, s1, s2, s3, h0, h1, h2, h3, 1, hoff);
452 k = hoff + 2 * step * len4;
454 d2 = dst + len4 + 2 + len2;
455 for (i = 0; i < (len4 - 2) >> 1; i++) {
456 CMUL2(d1, s0, s1, s2, s3, t0, t1, t2, t3, j, k);
457 CMUL2(d2, s0, s1, s2, s3, h0, h1, h2, h3, j, k);
461 CMUL0(dst, len2 + 4, s0, s1, s2, s3, t0, t1, t2, t3, 0, k);
465 float *tmp0,
float *tmp1)
467 memcpy(src, tmp0, 384 *
sizeof(*tmp0));
468 memcpy(tmp0 + 384, src + 384, 128 *
sizeof(*tmp0));
483 combine_fft(src, src + 128, src + 256, src + 384, tmp1,
494 memcpy(src, tmp1, 512 *
sizeof(
float));
498 float *tmp0,
float *tmp1)
500 memcpy(src, tmp0, 768 *
sizeof(*tmp0));
501 memcpy(tmp0 + 768, src + 768, 256 *
sizeof(*tmp0));
516 combine_fft(src, src + 256, src + 512, src + 768, tmp1,
527 memcpy(src, tmp1, 1024 *
sizeof(
float));
532 float *tmp0 = c->
temp, *tmp1 = c->
temp + 1024;
534 memset(tmp0, 0,
sizeof(*tmp0) * 1024);
535 memset(tmp1, 0,
sizeof(*tmp1) * 1024);
559 memset(tmp0, 0, 64 *
sizeof(*tmp0));
597 memset(tmp0, 0, 128 *
sizeof(*tmp0));
618 float *tmp0 = c->
temp, *tmp1 = c->
temp + 1024;
620 memset(tmp0, 0,
sizeof(*tmp0) * 1024);
621 memset(tmp1, 0,
sizeof(*tmp1) * 1024);
641 memset(tmp0, 0, 64 *
sizeof(*tmp0));
673 memset(tmp0, 0, 128 *
sizeof(*tmp0));
697 float *saved = c->
delay[ch];
699 float *wout = out + 448;
706 c->
wtf(c, buf,
in, 1024);
709 c->
wtf(c, buf,
in, 512);
711 for (i = 0; i < 256; i++) {
712 FFSWAP(
float, buf[i + 512], buf[1023 - i]);
717 for (i = 0; i < 256; i++) {
718 FFSWAP(
float, buf[i], buf[511 - i]);
720 c->
wtf(c, buf + 512,
in + 512, 512);
724 memcpy(out, saved, 448 *
sizeof(
float));
726 memcpy(wout + 128, buf + 64, 448 *
sizeof(
float));
727 memcpy(saved, buf + 512, 448 *
sizeof(
float));
728 memcpy(saved + 448, buf + 7*128 + 64, 64 *
sizeof(
float));
741 float *saved = c->
delay[channel];
763 float *wout = out + 448;
764 memcpy(out, saved, 448 *
sizeof(
float));
772 memcpy(wout + 4*128,
temp, 64 *
sizeof(
float));
775 memcpy(wout + 128, buf + 64, 448 *
sizeof(
float));
782 memcpy(saved,
temp + 64, 64 *
sizeof(
float));
786 memcpy(saved + 448, buf + 7*128 + 64, 64 *
sizeof(
float));
789 memcpy(saved, buf + 512, 448 *
sizeof(
float));
790 memcpy(saved + 448, buf + 7*128 + 64, 64 *
sizeof(
float));
794 memcpy(saved, buf + 512, 512 *
sizeof(
float));
842 int *got_frame_ptr,
AVPacket *avpkt)
846 int buf_size = avpkt->
size;
887 frame, audio_off)) < 0)
904 for (i = 1; i < 16; i++)
929 "Stereo mode support is not good, patch is welcome\n");
934 for (i = 0; i < 20; i++)
968 for (i = 1; i < 16; i++) {
972 syms, 2, 2, 0, 0, avctx);
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static int on2avc_reconstruct_channel_ext(On2AVCContext *c, AVFrame *dst, int offset)
const double *const ff_on2avc_tabs_4_10_1[4]
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static int on2avc_read_channel_data(On2AVCContext *c, GetBitContext *gb, int ch)
const double ff_on2avc_tab_20_1[]
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
float coeffs[2][ON2AVC_SUBFRAME_SIZE]
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
float delay[2][ON2AVC_SUBFRAME_SIZE]
static av_cold int init(AVCodecContext *avctx)
const On2AVCMode ff_on2avc_modes_40[8]
#define avpriv_request_sample(...)
static void on2avc_read_ms_info(On2AVCContext *c, GetBitContext *gb)
#define CMUL0(dst, id, s0, s1, s2, s3, t0, t1, t2, t3, is, it)
const double ff_on2avc_tab_84_4[]
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
static __device__ float ceil(float a)
#define AV_CH_LAYOUT_STEREO
float mdct_buf[ON2AVC_SUBFRAME_SIZE]
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static int on2avc_apply_ms(On2AVCContext *c)
static float on2avc_scale(int v, float scale)
static int on2avc_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static const uint8_t run_len[7][16]
const float ff_on2avc_window_short[128]
uint8_t band_type[ON2AVC_MAX_BANDS]
enum AVSampleFormat sample_fmt
audio sample format
static void wtf_end_1024(On2AVCContext *c, float *out, float *src, float *tmp0, float *tmp1)
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
static av_cold int on2avc_decode_init(AVCodecContext *avctx)
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But first
const double ff_on2avc_tab_84_1[]
const float ff_on2avc_ctab_1[2048]
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
static int on2avc_decode_band_types(On2AVCContext *c, GetBitContext *gb)
const double *const ff_on2avc_tabs_20_84_1[20]
bitstream reader API header.
static void wtf_40(On2AVCContext *c, float *out, float *src, int size)
uint8_t band_run_end[ON2AVC_MAX_BANDS]
const double ff_on2avc_tab_84_2[]
const double ff_on2avc_tab_40_2[]
static av_always_inline int bytestream2_get_bytes_left(GetByteContext *g)
static int on2avc_decode_subframe(On2AVCContext *c, const uint8_t *buf, int buf_size, AVFrame *dst, int offset)
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
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 uint16_t mask[17]
const double *const ff_on2avc_tabs_9_20_2[9]
static av_always_inline void bytestream2_skip(GetByteContext *g, unsigned int size)
#define CMUL1(dst, s0, s1, s2, s3, t0, t1, t2, t3, is, it)
#define CMUL2(dst, s0, s1, s2, s3, t0, t1, t2, t3, is, it)
const float ff_on2avc_window_long_24000[1024]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int flags
AV_CODEC_FLAG_*.
const char * name
Name of the codec implementation.
const double ff_on2avc_tab_20_2[]
const uint8_t ff_on2avc_scale_diff_syms[ON2AVC_SCALE_DIFFS]
const double ff_on2avc_tab_40_1[]
uint64_t channel_layout
Audio channel layout.
static int get_egolomb(GetBitContext *gb)
const double *const ff_on2avc_tabs_9_20_1[9]
static av_cold void on2avc_free_vlcs(On2AVCContext *c)
audio channel layout utility functions
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
static int on2avc_reconstruct_channel(On2AVCContext *c, int channel, AVFrame *dst, int offset)
static void wtf_end_512(On2AVCContext *c, float *out, float *src, float *tmp0, float *tmp1)
int ms_info[ON2AVC_MAX_BANDS]
float short_win[ON2AVC_SUBFRAME_SIZE/8]
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
void(* wtf)(struct On2AVCContext *ctx, float *out, float *in, int size)
const uint8_t ff_on2avc_cb_lens[]
static void twiddle(float *src1, float *src2, int src2_len, const double *tab, int tab_len, int step, int order0, int order1, const double *const *tabs)
static void pretwiddle(float *src, float *dst, int dst_len, int tab_step, int step, int order0, int order1, const double *const *tabs)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
const On2AVCMode ff_on2avc_modes_44[8]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
float long_win[ON2AVC_SUBFRAME_SIZE]
#define ON2AVC_SUBFRAME_SIZE
const double *const ff_on2avc_tabs_4_10_2[4]
const uint8_t ff_on2avc_scale_diff_bits[ON2AVC_SCALE_DIFFS]
Libavcodec external API header.
static int on2avc_decode_quads(On2AVCContext *c, GetBitContext *gb, float *dst, int dst_size, int type, float band_scale)
AVSampleFormat
Audio sample formats.
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.
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
const float ff_on2avc_ctab_2[2048]
static unsigned int get_bits1(GetBitContext *s)
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
AVCodec ff_on2avc_decoder
int ff_init_vlc_from_lengths(VLC *vlc_arg, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags, void *logctx)
Build VLC decoding tables suitable for use with get_vlc2()
const double ff_on2avc_tab_10_2[]
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
const double *const ff_on2avc_tabs_20_84_4[20]
static av_const int sign_extend(int val, unsigned bits)
const int ff_on2avc_cb_elems[]
const float ff_on2avc_ctab_4[2048]
internal math functions header
common internal api header.
static av_cold int on2avc_decode_close(AVCodecContext *avctx)
channel
Use these values when setting the channel map with ebur128_set_channel().
const double *const ff_on2avc_tabs_20_84_2[20]
void(* fft_calc)(struct FFTContext *s, FFTComplex *z)
Do a complex FFT with the parameters defined in ff_fft_init().
static void zero_head_and_tail(float *src, int len, int order0, int order1)
const double ff_on2avc_tab_84_3[]
float temp[ON2AVC_SUBFRAME_SIZE *2]
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
static int on2avc_decode_pairs(On2AVCContext *c, GetBitContext *gb, float *dst, int dst_size, int type, float band_scale)
const float ff_on2avc_window_long_32000[1024]
const double *const ff_on2avc_tabs_19_40_1[19]
static uint8_t half(int a, int b)
static const struct twinvq_data tab
const double *const ff_on2avc_tabs_19_40_2[19]
const double *const ff_on2avc_tabs_20_84_3[20]
static int on2avc_decode_band_scales(On2AVCContext *c, GetBitContext *gb)
static enum AVSampleFormat sample_fmts[]
const float ff_on2avc_ctab_3[2048]
const uint16_t ff_on2avc_cb_syms[]
static void wtf_44(On2AVCContext *c, float *out, float *src, int size)
#define ON2AVC_SCALE_DIFFS
static void combine_fft(float *s0, float *s1, float *s2, float *s3, float *dst, const float *t0, const float *t1, const float *t2, const float *t3, int len, int step)
#define FFSWAP(type, a, b)
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
static double val(void *priv, double ch)
This structure stores compressed data.
void ff_free_vlc(VLC *vlc)
static const struct @94 tabs[]
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators...
float band_scales[ON2AVC_MAX_BANDS]
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about which is also called distortion Distortion can be quantified by almost any quality measurement one chooses the sum of squared differences is used but more complex methods that consider psychovisual effects can be used as well It makes no difference in this discussion First step
const double ff_on2avc_tab_10_1[]