33 #define WEIGHT_LUT_NBITS 20
34 #define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
36 #define SQR(x) ((x) * (x))
74 #define OFFSET(x) offsetof(AudioNLMeansContext, x)
75 #define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
124 for (
int k = -K; k <= K; k++)
131 ptrdiff_t
S, ptrdiff_t K,
132 ptrdiff_t
i, ptrdiff_t jj)
136 for (
int j = jj; j < jj +
S; j++, v++)
137 cache[v] += -
SQR(
f[
i - K - 1] -
f[j - K - 1]) +
SQR(
f[
i + K] -
f[j + K]);
153 int newK, newS, newH, newN;
160 newN = newH + (newK + newS) * 2;
164 if (!
s->cache ||
s->cache->nb_samples < newS * 2) {
168 s->cache = new_cache;
178 float w = -
i /
s->pdiff_lut_scale;
183 if (!
s->in ||
s->in->nb_samples < newN) {
235 const int om =
s->om;
236 const float *
f = (
const float *)(
s->in->extended_data[ch]) + K;
237 float *cache = (
float *)
s->cache->extended_data[ch];
238 const float sw = (65536.f / (4 * K + 2)) / sqrtf(
s->a);
239 float *dst = (
float *)
out->extended_data[ch] +
s->offset;
243 float P = 0.f, Q = 0.f;
247 for (
int j =
i -
S; j <=
i +
S; j++) {
250 cache[v++] =
s->dsp.compute_distance_ssd(
f +
i,
f + j, K);
253 s->dsp.compute_cache(cache,
f,
S, K,
i,
i -
S);
254 s->dsp.compute_cache(cache +
S,
f,
S, K,
i,
i + 1);
257 for (
int j = 0; j < 2 *
S && !
ctx->is_disabled; j++) {
259 unsigned weight_lut_idx;
269 weight_lut_idx =
w *
s->pdiff_lut_scale;
271 w =
s->weight_lut[weight_lut_idx];
272 P +=
w *
f[
i -
S + j + (j >=
S)];
295 int available, wanted, ret;
306 wanted = (available /
s->H) *
s->H;
308 if (wanted >=
s->H && available >=
s->N) {
314 while (available >=
s->N) {
329 out->nb_samples =
s->offset;
330 if (
s->eof_left >= 0) {
331 out->nb_samples =
FFMIN(
s->eof_left,
s->offset);
332 s->eof_left -=
out->nb_samples;
355 if (
s->eof_left <= 0)
368 char *res,
int res_len,
int flags)
413 .description =
NULL_IF_CONFIG_SMALL(
"Reduce broadband noise from stream using Non-Local Means."),
416 .priv_class = &anlmdn_class,
static enum AVSampleFormat sample_fmts[]
static int config_filter(AVFilterContext *ctx)
AVFILTER_DEFINE_CLASS(anlmdn)
static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad inputs[]
static int request_frame(AVFilterLink *outlink)
static const AVFilterPad outputs[]
static const AVOption anlmdn_options[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
static void compute_cache_c(float *cache, const float *f, ptrdiff_t S, ptrdiff_t K, ptrdiff_t i, ptrdiff_t jj)
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static av_cold void uninit(AVFilterContext *ctx)
static int config_output(AVFilterLink *outlink)
void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
void ff_anlmdn_init_x86(AudioNLMDNDSPContext *s)
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().
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
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.
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Main libavfilter public API header.
#define flags(name, subs,...)
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
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.
#define AVERROR_EOF
End of file.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
#define AV_NOPTS_VALUE
Undefined timestamp value.
#define AV_TIME_BASE
Internal time base represented as integer.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
static float distance(float x, float y, int band)
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
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
int sample_rate
samples per second
AVFilterContext * dst
dest 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.
AVFormatInternal * internal
An opaque field for libavformat internal usage.
This structure describes decoded (raw) audio or video data.
Rational number (pair of numerator and denominator).
float(* compute_distance_ssd)(const float *f1, const float *f2, ptrdiff_t K)
void(* compute_cache)(float *cache, const float *f, ptrdiff_t S, ptrdiff_t K, ptrdiff_t i, ptrdiff_t jj)
float weight_lut[WEIGHT_LUT_SIZE]
static float smooth(DeshakeOpenCLContext *deshake_ctx, float *gauss_kernel, int length, float max_val, AVFifoBuffer *values)