63 #define OFFSET(x) offsetof(ChorusContext, x)
64 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
83 for (p = item_str; *p; p++) {
90 static void fill_items(
char *item_str,
int *nb_items,
float *items)
92 char *p, *saveptr =
NULL;
93 int i, new_nb_items = 0;
96 for (
i = 0;
i < *nb_items;
i++) {
100 new_nb_items += sscanf(tstr,
"%f", &items[new_nb_items]) == 1;
103 *nb_items = new_nb_items;
109 int nb_delays, nb_decays, nb_speeds, nb_depths;
111 if (!
s->delays_str || !
s->decays_str || !
s->speeds_str || !
s->depths_str) {
126 if (!
s->delays || !
s->decays || !
s->speeds || !
s->depths)
134 if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
139 s->num_chorus = nb_delays;
141 if (
s->num_chorus < 1) {
146 s->length =
av_calloc(
s->num_chorus,
sizeof(*
s->length));
147 s->lookup_table =
av_calloc(
s->num_chorus,
sizeof(*
s->lookup_table));
149 if (!
s->length || !
s->lookup_table)
190 float sum_in_volume = 1.0;
195 for (n = 0; n <
s->num_chorus; n++) {
196 int samples = (
int) ((
s->delays[n] +
s->depths[n]) * outlink->
sample_rate / 1000.0);
197 int depth_samples = (
int) (
s->depths[n] * outlink->
sample_rate / 1000.0);
202 if (!
s->lookup_table[n])
206 s->length[n], 0., depth_samples, 0);
207 s->max_samples =
FFMAX(
s->max_samples, samples);
210 for (n = 0; n <
s->num_chorus; n++)
211 sum_in_volume +=
s->decays[n];
213 if (
s->in_gain * (sum_in_volume) > 1.0 /
s->out_gain)
224 for (n = 0; n < outlink->
channels; n++) {
225 s->phase[n] =
av_calloc(
s->num_chorus,
sizeof(
int));
230 s->fade_out =
s->max_samples;
238 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
261 float *chorusbuf = (
float *)
s->chorusbuf[
c];
262 int *phase =
s->phase[
c];
269 for (n = 0; n <
s->num_chorus; n++) {
270 out += chorusbuf[
MOD(
s->max_samples +
s->counter[
c] -
271 s->lookup_table[n][phase[n]],
272 s->max_samples)] *
s->decays[n];
273 phase[n] =
MOD(phase[n] + 1,
s->length[n]);
280 chorusbuf[
s->counter[
c]] =
in;
281 s->counter[
c] =
MOD(
s->counter[
c] + 1,
s->max_samples);
287 if (
frame != out_frame)
302 int nb_samples =
FFMIN(
s->fade_out, 2048);
308 s->fade_out -= nb_samples;
340 for (n = 0; n <
s->channels; n++)
348 for (n = 0; n <
s->num_chorus; n++)
377 .priv_class = &chorus_class,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static int query_formats(AVFilterContext *ctx)
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
static int request_frame(AVFilterLink *outlink)
static const AVFilterPad chorus_inputs[]
static const AVFilterPad chorus_outputs[]
AVFILTER_DEFINE_CLASS(chorus)
static av_cold int init(AVFilterContext *ctx)
static av_cold void uninit(AVFilterContext *ctx)
static const AVOption chorus_options[]
static int config_output(AVFilterLink *outlink)
static void count_items(char *item_str, int *nb_items)
static void fill_items(char *item_str, int *nb_items, float *items)
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
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.
void ff_generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, void *table, int table_size, double min, double max, double phase)
#define AVERROR_EOF
End of file.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
#define AV_LOG_WARNING
Something somehow does not look correct.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
@ AV_SAMPLE_FMT_S32
signed 32 bits
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a data pointers array, samples buffer for nb_samples samples, and fill data pointers and lin...
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().
#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
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.
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).
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
uint8_t ** extended_data
pointers to the data planes/channels.
Rational number (pair of numerator and denominator).
#define av_realloc_f(p, o, n)