57 static const char *
const var_names[] = {
"sr",
"b",
"nb",
"ch",
"chs",
"pts",
"re",
"im",
NULL };
60 #define OFFSET(x) offsetof(AFFTFiltContext, x) 61 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 95 static inline double getreal(
void *priv,
double x,
double ch)
100 ich = av_clip(ch, 0, s->
nb_exprs - 1);
106 static inline double getimag(
void *priv,
double x,
double ch)
111 ich = av_clip(ch, 0, s->
nb_exprs - 1);
117 static double realf(
void *priv,
double x,
double ch) {
return getreal(priv, x, ch); }
118 static double imagf(
void *priv,
double x,
double ch) {
return getimag(priv, x, ch); }
127 char *saveptr =
NULL;
131 const char *last_expr =
"1";
151 for (ch = 0; ch < inlink->
channels; ch++) {
157 for (ch = 0; ch < inlink->
channels; ch++) {
175 for (ch = 0; ch < inlink->
channels; ch++) {
195 for (ch = 0; ch < inlink->
channels; ch++) {
255 for (ch = 0; ch < inlink->
channels; ch++) {
275 for (ch = 0; ch < inlink->
channels; ch++) {
282 for (ch = 0; ch < inlink->
channels; ch++) {
289 for (n = 0; n <= window_size / 2; n++) {
303 for (n = window_size / 2 + 1, x = window_size / 2 - 1; n <
window_size; n++, x--) {
304 fft_temp[n].
re = fft_temp[x].
re;
305 fft_temp[n].
im = -fft_temp[x].
im;
325 for (ch = 0; ch < inlink->
channels; ch++) {
330 dst[n] = buf[n] * (1.f - s->
overlap);
331 memmove(buf, buf + s->
hop_size, window_size * 4);
342 return ret < 0 ? ret : 0;
481 .description =
NULL_IF_CONFIG_SMALL(
"Apply arbitrary expressions to samples in frequency domain."),
483 .priv_class = &afftfilt_class,
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
#define av_realloc_f(p, o, n)
This structure describes decoded (raw) audio or video data.
av_cold void av_fft_end(FFTContext *s)
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
static int config_input(AVFilterLink *inlink)
static const char *const var_names[]
static double realf(void *priv, double x, double ch)
#define FFERROR_NOT_READY
Filters implementation helper functions.
void av_fft_permute(FFTContext *s, FFTComplex *z)
Do the permutation needed BEFORE calling ff_fft_calc().
static void generate_window_func(float *lut, int N, int win_func, float *overlap)
int av_expr_parse(AVExpr **expr, const char *s, const char *const *const_names, const char *const *func1_names, double(*const *funcs1)(void *, double), const char *const *func2_names, double(*const *funcs2)(void *, double, double), int log_offset, void *log_ctx)
Parse an expression.
static int filter_frame(AVFilterLink *inlink)
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 * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
static av_cold void uninit(AVFilterContext *ctx)
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
#define AVERROR_EOF
End of file.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
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.
static double imagf(void *priv, double x, double ch)
#define i(width, name, range_min, range_max)
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.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void * priv
private data for use by the filter
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
FFTContext * av_fft_init(int nbits, int inverse)
Set up a complex FFT.
Context for an Audio FIFO Buffer.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad outputs[]
int format
agreed upon media format
A list of supported channel layouts.
static double getreal(void *priv, double x, double ch)
static const AVFilterPad inputs[]
char * av_strdup(const char *s)
Duplicate a string.
AVSampleFormat
Audio sample formats.
void av_expr_free(AVExpr *e)
Free a parsed expression previously created with av_expr_parse().
double(* func2[])(void *, double, double)
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
Describe the class of an AVClass context structure.
Rational number (pair of numerator and denominator).
static const AVOption afftfilt_options[]
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok()...
common internal and external API header
void ff_filter_set_ready(AVFilterContext *filter, unsigned priority)
Mark a filter ready and schedule it for activation.
int channels
Number of channels.
AVFILTER_DEFINE_CLASS(afftfilt)
static const char *const func2_names[]
static double getimag(void *priv, double x, double ch)
static int activate(AVFilterContext *ctx)
double av_expr_eval(AVExpr *e, const double *const_values, void *opaque)
Evaluate a previously parsed expression.
AVFilterContext * dst
dest filter
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
Peek data from an AVAudioFifo.
static enum AVSampleFormat sample_fmts[]
uint8_t ** extended_data
pointers to the data planes/channels.
void av_fft_calc(FFTContext *s, FFTComplex *z)
Do a complex FFT with the parameters defined in av_fft_init().
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_NOPTS_VALUE
Undefined timestamp value.
simple arithmetic expression evaluator