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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)
161 float sum_in_volume = 1.0;
166 for (n = 0; n <
s->num_chorus; n++) {
168 int depth_samples = (
int) (
s->depths[n] * outlink->
sample_rate / 1000.0);
173 if (!
s->lookup_table[n])
177 s->length[n], 0., depth_samples, 0);
181 for (n = 0; n <
s->num_chorus; n++)
182 sum_in_volume +=
s->decays[n];
184 if (
s->in_gain * (sum_in_volume) > 1.0 /
s->out_gain)
196 s->phase[n] =
av_calloc(
s->num_chorus,
sizeof(
int));
201 s->fade_out =
s->max_samples;
209 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
229 for (
c = 0;
c <
inlink->ch_layout.nb_channels;
c++) {
230 const float *
src = (
const float *)
frame->extended_data[
c];
232 float *chorusbuf = (
float *)
s->chorusbuf[
c];
233 int *phase =
s->phase[
c];
238 out = in *
s->in_gain;
240 for (n = 0; n <
s->num_chorus; n++) {
241 out += chorusbuf[
MOD(
s->max_samples +
s->counter[
c] -
242 s->lookup_table[n][phase[n]],
243 s->max_samples)] *
s->decays[n];
244 phase[n] =
MOD(phase[n] + 1,
s->length[n]);
251 chorusbuf[
s->counter[
c]] = in;
252 s->counter[
c] =
MOD(
s->counter[
c] + 1,
s->max_samples);
258 if (
frame != out_frame)
273 int nb_samples =
FFMIN(
s->fade_out, 2048);
279 s->fade_out -= nb_samples;
311 for (n = 0; n <
s->channels; n++)
319 for (n = 0; n <
s->num_chorus; n++)
345 .priv_class = &chorus_class,
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
@ AV_SAMPLE_FMT_FLTP
float, planar
#define AV_LOG_WARNING
Something somehow does not look correct.
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
#define AVERROR_EOF
End of file.
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
This structure describes decoded (raw) audio or video data.
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
static av_cold void uninit(AVFilterContext *ctx)
const char * name
Filter name.
int nb_channels
Number of channels in this layout.
A link between two filters.
static void fill_items(char *item_str, int *nb_items, float *items)
static void count_items(char *item_str, int *nb_items)
A filter pad used for either input or output.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
AVFILTER_DEFINE_CLASS(chorus)
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().
const AVFilter ff_af_chorus
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
#define FILTER_INPUTS(array)
#define av_realloc_f(p, o, n)
Describe the class of an AVClass context structure.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Rational number (pair of numerator and denominator).
static int config_output(AVFilterLink *outlink)
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
void ff_generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, void *table, int table_size, double min, double max, double phase)
static const AVFilterPad chorus_inputs[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
int format
agreed upon media format
#define AV_NOPTS_VALUE
Undefined timestamp value.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
AVFilterContext * src
source filter
int sample_rate
samples per second
#define i(width, name, range_min, range_max)
static const AVFilterPad chorus_outputs[]
static const AVOption chorus_options[]
uint8_t ** extended_data
pointers to the data planes/channels.
const char * name
Pad name.
void * av_calloc(size_t nmemb, size_t size)
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.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static int request_frame(AVFilterLink *outlink)
Filter the word “frame” indicates either a video frame or a group of audio samples
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
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...
static av_cold int init(AVFilterContext *ctx)
AVChannelLayout ch_layout
channel layout of current buffer (see libavutil/channel_layout.h)
#define FILTER_OUTPUTS(array)
@ AV_SAMPLE_FMT_S32
signed 32 bits