100 static float xcorrelate(
const float *x,
const float *y,
float sumx,
float sumy,
int size)
102 const float xm = sumx /
size, ym = sumy /
size;
103 float num = 0.f, den, den0 = 0.f, den1 = 0.f;
105 for (
int i = 0;
i <
size;
i++) {
106 float xd = x[
i] - xm;
107 float yd = y[
i] - ym;
115 den = sqrtf((den0 * den1) / (
size *
size));
117 return den <= 1e-6f ? 0.f : num / den;
123 const int size =
s->size;
126 for (
int ch = 0; ch <
out->channels; ch++) {
127 const float *x = (
const float *)
s->cache[0]->extended_data[ch];
128 const float *y = (
const float *)
s->cache[1]->extended_data[ch];
129 float *sumx = (
float *)
s->mean_sum[0]->extended_data[ch];
130 float *sumy = (
float *)
s->mean_sum[1]->extended_data[ch];
131 float *dst = (
float *)
out->extended_data[ch];
140 for (
int n = 0; n <
out->nb_samples; n++) {
144 sumx[0] += x[n +
size];
146 sumy[0] += y[n +
size];
156 const int size =
s->size;
159 for (
int ch = 0; ch <
out->channels; ch++) {
160 const float *x = (
const float *)
s->cache[0]->extended_data[ch];
161 const float *y = (
const float *)
s->cache[1]->extended_data[ch];
162 float *num_sum = (
float *)
s->num_sum->extended_data[ch];
163 float *den_sumx = (
float *)
s->den_sum[0]->extended_data[ch];
164 float *den_sumy = (
float *)
s->den_sum[1]->extended_data[ch];
165 float *dst = (
float *)
out->extended_data[ch];
175 for (
int n = 0; n <
out->nb_samples; n++) {
178 num = num_sum[0] /
size;
179 den = sqrtf((den_sumx[0] * den_sumy[0]) / (
size *
size));
181 dst[n] = den <= 1e-6f ? 0.f : num / den;
183 num_sum[0] -= x[n] * y[n];
184 num_sum[0] += x[n +
size] * y[n +
size];
185 den_sumx[0] -= x[n] * x[n];
186 den_sumx[0] =
FFMAX(den_sumx[0], 0.f);
187 den_sumx[0] += x[n +
size] * x[n +
size];
188 den_sumy[0] -= y[n] * y[n];
189 den_sumy[0] =
FFMAX(den_sumy[0], 0.f);
190 den_sumy[0] += y[n +
size] * y[n +
size];
207 for (
int i = 0;
i < 2;
i++) {
221 if (available >
s->size) {
222 const int out_samples = available -
s->size;
225 if (!
s->cache[0] ||
s->cache[0]->nb_samples < available) {
232 if (!
s->cache[1] ||
s->cache[1]->nb_samples < available) {
254 s->pts += out_samples;
268 for (
int i = 0;
i < 2;
i++) {
276 for (
int i = 0;
i < 2;
i++) {
299 if (!
s->fifo[0] || !
s->fifo[1])
307 if (!
s->mean_sum[0] || !
s->mean_sum[1] || !
s->num_sum ||
308 !
s->den_sum[0] || !
s->den_sum[1])
336 .
name =
"axcorrelate0",
340 .name =
"axcorrelate1",
355 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
356 #define OFFSET(x) offsetof(AudioXCorrelateContext, x)
369 .
name =
"axcorrelate",
372 .priv_class = &axcorrelate_class,
static enum AVSampleFormat sample_fmts[]
AVFilter ff_af_axcorrelate
static float mean_sum(const float *in, int size)
static int query_formats(AVFilterContext *ctx)
static int xcorrelate_slow(AVFilterContext *ctx, AVFrame *out)
static int xcorrelate_fast(AVFilterContext *ctx, AVFrame *out)
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static float square_sum(const float *x, const float *y, int size)
AVFILTER_DEFINE_CLASS(axcorrelate)
static int activate(AVFilterContext *ctx)
static av_cold void uninit(AVFilterContext *ctx)
static const AVOption axcorrelate_options[]
static float xcorrelate(const float *x, const float *y, float sumx, float sumy, int size)
static int config_output(AVFilterLink *outlink)
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
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
simple assert() macros that are a bit more flexible than ISO C assert().
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Main libavfilter public API header.
audio channel layout utility functions
common internal and external API header
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
#define FFERROR_NOT_READY
Filters implementation helper functions.
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
Peek data from an AVAudioFifo.
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
#define AV_NOPTS_VALUE
Undefined timestamp value.
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
Context for an Audio FIFO Buffer.
Describe the class of an AVClass context structure.
A list of supported channel layouts.
A link between two filters.
int channels
Number of channels.
AVFilterContext * src
source filter
int format
agreed upon media format
A filter pad used for either input or output.
const char * name
Pad name.
const char * name
Filter name.
This structure describes decoded (raw) audio or video data.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
uint8_t ** extended_data
pointers to the data planes/channels.
int(* xcorrelate)(AVFilterContext *ctx, AVFrame *out)