43 static void fcmul_add_c(
float *sum,
const float *t,
const float *
c, ptrdiff_t
len)
47 for (n = 0; n <
len; n++) {
48 const float cre = c[2 * n ];
49 const float cim = c[2 * n + 1];
50 const float tre = t[2 * n ];
51 const float tim = t[2 * n + 1];
53 sum[2 * n ] += tre * cre - tim * cim;
54 sum[2 * n + 1] += tre * cim + tim * cre;
57 sum[2 * n] += t[2 * n] * c[2 * n];
62 for (
int n = 0; n <
len; n++)
63 for (
int m = 0; m <= n; m++)
64 out[n] += ir[m].
re * in[n - m];
85 for (n = 0; n < nb_samples; n++)
96 for (n = 0; n < nb_samples; n++) {
111 direct(src, coeff, nb_samples, dst);
122 for (n = 0; n < nb_samples; n++) {
128 memset(sum, 0,
sizeof(*sum) * seg->
fft_length);
132 memcpy(block, src,
sizeof(*src) * seg->
part_size);
160 memcpy(dst, buf, seg->
part_size *
sizeof(*dst));
169 for (n = 0; n < nb_samples; n++) {
178 for (n = 0; n < nb_samples; n++)
199 const int start = (out->
channels * jobnr) / nb_jobs;
200 const int end = (out->
channels * (jobnr+1)) / nb_jobs;
202 for (
int ch = start; ch <
end; ch++) {
244 for (i = 0; txt[
i]; i++) {
248 for (char_y = 0; char_y < font_height; char_y++) {
249 for (mask = 0x80;
mask; mask >>= 1) {
250 if (font[txt[i] * font_height + char_y] & mask)
261 int dx =
FFABS(x1-x0);
262 int dy =
FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
263 int err = (dx>dy ? dx : -dy) / 2, e2;
268 if (x0 == x1 && y0 == y1)
288 float *mag, *phase, *delay,
min = FLT_MAX,
max = FLT_MIN;
289 float min_delay = FLT_MAX, max_delay = FLT_MIN;
290 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
299 if (!mag || !phase || !delay)
303 for (i = 0; i < s->
w; i++) {
305 double w = i *
M_PI / (s->
w - 1);
306 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
308 for (x = 0; x < s->
nb_taps; x++) {
309 real += cos(-x *
w) * src[x];
310 imag += sin(-x * w) * src[x];
311 real_num += cos(-x * w) * src[x] * x;
312 imag_num += sin(-x * w) * src[x] * x;
315 mag[
i] =
hypot(real, imag);
316 phase[
i] = atan2(imag, real);
317 div = real * real + imag * imag;
318 delay[
i] = (real_num * real + imag_num * imag) / div;
319 min =
fminf(min, mag[i]);
320 max =
fmaxf(max, mag[i]);
321 min_delay =
fminf(min_delay, delay[i]);
322 max_delay =
fmaxf(max_delay, delay[i]);
325 for (i = 0; i < s->
w; i++) {
326 int ymag = mag[
i] / max * (s->
h - 1);
327 int ydelay = (delay[
i] - min_delay) / (max_delay - min_delay) * (s->
h - 1);
328 int yphase = (0.5 * (1. + phase[
i] /
M_PI)) * (s->
h - 1);
330 ymag = s->
h - 1 - av_clip(ymag, 0, s->
h - 1);
331 yphase = s->
h - 1 - av_clip(yphase, 0, s->
h - 1);
332 ydelay = s->
h - 1 - av_clip(ydelay, 0, s->
h - 1);
337 prev_yphase = yphase;
339 prev_ydelay = ydelay;
342 draw_line(out, i, yphase,
FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
343 draw_line(out, i, ydelay,
FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
346 prev_yphase = yphase;
347 prev_ydelay = ydelay;
350 if (s->
w > 400 && s->
h > 100) {
351 drawtext(out, 2, 2,
"Max Magnitude:", 0xDDDDDDDD);
352 snprintf(text,
sizeof(text),
"%.2f", max);
353 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
355 drawtext(out, 2, 12,
"Min Magnitude:", 0xDDDDDDDD);
356 snprintf(text,
sizeof(text),
"%.2f", min);
357 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
359 drawtext(out, 2, 22,
"Max Delay:", 0xDDDDDDDD);
360 snprintf(text,
sizeof(text),
"%.2f", max_delay);
361 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
363 drawtext(out, 2, 32,
"Min Delay:", 0xDDDDDDDD);
364 snprintf(text,
sizeof(text),
"%.2f", min_delay);
365 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
375 int offset,
int nb_partitions,
int part_size)
397 for (
int ch = 0; ch < ctx->
inputs[0]->
channels && part_size >= 8; ch++) {
449 int ret,
i, ch, n, cur_nb_taps;
453 int part_size, max_part_size;
470 for (i = 0; left > 0; i++) {
471 int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
472 int nb_partitions =
FFMIN(step, (left + part_size - 1) / part_size);
478 offset += nb_partitions * part_size;
479 left -= nb_partitions * part_size;
481 part_size =
FFMIN(part_size, max_part_size);
507 for (i = 0; i < cur_nb_taps; i++)
508 power +=
FFABS(time[i]);
516 for (i = 0; i < cur_nb_taps; i++)
525 for (i = 0; i < cur_nb_taps; i++)
526 power += time[i] * time[i];
528 s->
gain = sqrtf(ch / power);
562 const float scale = 1.f / seg->
part_size;
564 const int remaining = s->
nb_taps - toffset;
568 for (n = 0; n <
size; n++)
569 coeff[coffset + n].
re = time[toffset + n];
575 memset(block, 0,
sizeof(*block) * seg->
fft_length);
576 memcpy(block, time + toffset, size *
sizeof(*block));
580 coeff[coffset].re = block[0] * scale;
581 coeff[coffset].im = 0;
583 coeff[coffset + n].re = block[2 * n] * scale;
584 coeff[coffset + n].im = block[2 * n + 1] * scale;
611 int nb_taps, max_nb_taps;
615 if (nb_taps > max_nb_taps) {
616 av_log(ctx,
AV_LOG_ERROR,
"Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
627 int ret, status, available, wanted;
854 for (
int n = 0; n < s->
nb_irs; n++) {
918 int prev_ir = s->
selir;
926 if (prev_ir != s->
selir) {
933 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 934 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM 935 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 936 #define OFFSET(x) offsetof(AudioFIRContext, x) 953 {
"channel",
"set IR channel to display frequency response",
OFFSET(ir_channel),
AV_OPT_TYPE_INT, {.i64=0}, 0, 1024,
VF },
967 .description =
NULL_IF_CONFIG_SMALL(
"Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
969 .priv_class = &afir_class,
static av_cold int init(AVFilterContext *ctx)
This structure describes decoded (raw) audio or video data.
static int check_ir(AVFilterLink *link, AVFrame *frame)
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Main libavfilter public API header.
int h
agreed upon image height
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
static av_cold void uninit(AVFilterContext *ctx)
float fminf(float, float)
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int offset, int nb_partitions, int part_size)
#define FFERROR_NOT_READY
Filters implementation helper functions.
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
struct AVFilterChannelLayouts * in_channel_layouts
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
static void draw_response(AVFilterContext *ctx, AVFrame *out)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
AVFilterPad * output_pads
array of output pads
packed RGB 8:8:8, 32bpp, RGBXRGBX... X=unused/undefined
static av_cold int end(AVCodecContext *avctx)
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
void ff_afir_init_x86(AudioFIRDSPContext *s)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
#define AVERROR_EOF
End of file.
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
static int process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
A filter pad used for either input or output.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
A link between two filters.
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
AVFilterPad * input_pads
array of input pads
#define i(width, name, range_min, range_max)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
const uint8_t avpriv_cga_font[2048]
static int convert_coeffs(AVFilterContext *ctx)
AVFILTER_DEFINE_CLASS(afir)
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
AVRational frame_rate
Frame rate of the stream on the link, or 1/0 if unknown or variable; if left to 0/0, will be automatically copied from the first input of the source filter if it exists.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
static const uint16_t mask[17]
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
unsigned nb_outputs
number of output pads
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options...
void * priv
private data for use by the filter
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static void direct(const float *in, const FFTComplex *ir, int len, float *out)
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
struct AVFilterChannelLayouts * out_channel_layouts
AVFilterFormats * in_formats
Lists of formats and channel layouts supported by the input and output filters respectively.
static const uint8_t offset[127][2]
void av_rdft_calc(RDFTContext *s, FFTSample *data)
int w
agreed upon image width
char * av_asprintf(const char *fmt,...)
static av_const double hypot(double x, double y)
int channels
number of audio channels, only used for audio.
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
unsigned nb_inputs
number of input pads
float fmaxf(float, float)
int ff_inlink_queued_samples(AVFilterLink *link)
AudioFIRDSPContext afirdsp
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
void(* fcmul_add)(float *sum, const float *t, const float *c, ptrdiff_t len)
void av_rdft_end(RDFTContext *s)
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
AVFilterContext * src
source filter
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
void ff_afir_init(AudioFIRDSPContext *dsp)
static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
A list of supported channel layouts.
static int query_formats(AVFilterContext *ctx)
char * av_strdup(const char *s)
Duplicate a string.
AVSampleFormat
Audio sample formats.
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Rational number (pair of numerator and denominator).
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
offset must point to AVRational
const char * name
Filter name.
AVRational sample_aspect_ratio
agreed upon sample aspect ratio
offset must point to two consecutive integers
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
static enum AVPixelFormat pix_fmts[]
#define flags(name, subs,...)
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static av_always_inline AVRational av_inv_q(AVRational q)
Invert a rational.
common internal and external API header
AudioFIRSegment seg[1024]
channel
Use these values when setting the channel map with ebur128_set_channel().
static int config_video(AVFilterLink *outlink)
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
int channels
Number of channels.
avfilter_execute_func * execute
static const AVOption afir_options[]
static int activate(AVFilterContext *ctx)
AVFilterContext * dst
dest filter
static const double coeff[2][5]
static enum AVSampleFormat sample_fmts[]
#define av_malloc_array(a, b)
static int config_output(AVFilterLink *outlink)
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
AVPixelFormat
Pixel format.
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_NOPTS_VALUE
Undefined timestamp value.
CGA/EGA/VGA ROM font data.
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)