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29 #define INTERPOLATION_LINEAR 0
30 #define INTERPOLATION_QUADRATIC 1
52 #define OFFSET(x) offsetof(FlangerContext, x)
53 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
58 {
"regen",
"percentage regeneration (delayed signal feedback)",
OFFSET(feedback_gain),
AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95,
A },
59 {
"width",
"percentage of delayed signal mixed with original",
OFFSET(delay_gain),
AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100,
A },
66 {
"phase",
"swept wave percentage phase-shift for multi-channel",
OFFSET(channel_phase),
AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100,
A },
79 s->feedback_gain /= 100;
81 s->channel_phase /= 100;
83 s->delay_depth /= 1000;
84 s->in_gain = 1 / (1 +
s->delay_gain);
85 s->delay_gain /= 1 +
s->delay_gain;
86 s->delay_gain *= 1 -
fabs(
s->feedback_gain);
96 s->max_samples = (
s->delay_min +
s->delay_depth) *
inlink->sample_rate + 2.5;
97 s->lfo_length =
inlink->sample_rate /
s->speed;
100 if (!
s->lfo || !
s->delay_last)
105 s->max_samples - 2., 3 *
M_PI_2);
108 inlink->ch_layout.nb_channels,
s->max_samples,
130 for (
i = 0;
i <
frame->nb_samples;
i++) {
132 s->delay_buf_pos = (
s->delay_buf_pos +
s->max_samples - 1) %
s->max_samples;
134 for (chan = 0; chan <
inlink->ch_layout.nb_channels; chan++) {
135 double *
src = (
double *)
frame->extended_data[chan];
137 double delayed_0, delayed_1;
140 int channel_phase = chan *
s->lfo_length *
s->channel_phase + .5;
141 double delay =
s->lfo[(
s->lfo_pos + channel_phase) %
s->lfo_length];
142 int int_delay = (
int)delay;
143 double frac_delay = modf(delay, &delay);
144 double *delay_buffer = (
double *)
s->delay_buffer[chan];
147 delay_buffer[
s->delay_buf_pos] = in +
s->delay_last[chan] *
149 delayed_0 = delay_buffer[(
s->delay_buf_pos + int_delay++) %
s->max_samples];
150 delayed_1 = delay_buffer[(
s->delay_buf_pos + int_delay++) %
s->max_samples];
153 delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
156 double delayed_2 = delay_buffer[(
s->delay_buf_pos + int_delay++) %
s->max_samples];
157 delayed_2 -= delayed_0;
158 delayed_1 -= delayed_0;
159 a = delayed_2 * .5 - delayed_1;
160 b = delayed_1 * 2 - delayed_2 *.5;
161 delayed = delayed_0 + (
a * frac_delay +
b) * frac_delay;
164 s->delay_last[chan] = delayed;
165 out = in *
s->in_gain + delayed *
s->delay_gain;
168 s->lfo_pos = (
s->lfo_pos + 1) %
s->lfo_length;
171 if (
frame != out_frame)
209 .priv_class = &flanger_class,
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
static int config_input(AVFilterLink *inlink)
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 FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
static av_cold void uninit(AVFilterContext *ctx)
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 const AVOption flanger_options[]
#define INTERPOLATION_QUADRATIC
const char * name
Filter name.
A link between two filters.
A filter pad used for either input or output.
static av_cold int init(AVFilterContext *ctx)
#define FILTER_INPUTS(array)
Describe the class of an AVClass context structure.
static __device__ float fabs(float a)
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
static const AVFilterPad flanger_outputs[]
void ff_generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, void *table, int table_size, double min, double max, double phase)
const AVFilter ff_af_flanger
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
static int interpolation(DeclickChannel *c, const double *src, int ar_order, double *acoefficients, int *index, int nb_errors, double *auxiliary, double *interpolated)
#define i(width, name, range_min, range_max)
uint8_t ** extended_data
pointers to the data planes/channels.
AVFILTER_DEFINE_CLASS(flanger)
const char * name
Pad name.
void * av_calloc(size_t nmemb, size_t size)
#define INTERPOLATION_LINEAR
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 const AVFilterPad flanger_inputs[]
@ AV_SAMPLE_FMT_DBLP
double, planar
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...
#define FILTER_OUTPUTS(array)