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45 uint8_t *
const *
src, uint8_t **dst,
49 #define OFFSET(x) offsetof(AudioEchoContext, x)
50 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
67 for (p = item_str; *p; p++) {
74 static void fill_items(
char *item_str,
int *nb_items,
float *items)
76 char *p, *saveptr =
NULL;
77 int i, new_nb_items = 0;
80 for (
i = 0;
i < *nb_items;
i++) {
84 new_nb_items +=
av_sscanf(tstr,
"%f", &items[new_nb_items]) == 1;
87 *nb_items = new_nb_items;
106 int nb_delays, nb_decays,
i;
108 if (!
s->delays || !
s->decays) {
118 if (!
s->delay || !
s->decay)
124 if (nb_delays != nb_decays) {
125 av_log(
ctx,
AV_LOG_ERROR,
"Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
129 s->nb_echoes = nb_delays;
139 for (
i = 0;
i < nb_delays;
i++) {
140 if (
s->delay[
i] <= 0 ||
s->delay[
i] > 90000) {
144 if (
s->decay[
i] <= 0 ||
s->decay[
i] > 1) {
156 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
158 #define ECHO(name, type, min, max) \
159 static void echo_samples_## name ##p(AudioEchoContext *ctx, \
160 uint8_t **delayptrs, \
161 uint8_t * const *src, uint8_t **dst, \
162 int nb_samples, int channels) \
164 const double out_gain = ctx->out_gain; \
165 const double in_gain = ctx->in_gain; \
166 const int nb_echoes = ctx->nb_echoes; \
167 const int max_samples = ctx->max_samples; \
168 int i, j, chan, av_uninit(index); \
170 av_assert1(channels > 0); \
172 for (chan = 0; chan < channels; chan++) { \
173 const type *s = (type *)src[chan]; \
174 type *d = (type *)dst[chan]; \
175 type *dbuf = (type *)delayptrs[chan]; \
177 index = ctx->delay_index; \
178 for (i = 0; i < nb_samples; i++, s++, d++) { \
182 out = in * in_gain; \
183 for (j = 0; j < nb_echoes; j++) { \
184 int ix = index + max_samples - ctx->samples[j]; \
185 ix = MOD(ix, max_samples); \
186 out += dbuf[ix] * ctx->decay[j]; \
190 *d = av_clipd(out, min, max); \
193 index = MOD(index + 1, max_samples); \
196 ctx->delay_index = index; \
199 ECHO(dbl,
double, -1.0, 1.0 )
200 ECHO(flt,
float, -1.0, 1.0 )
201 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
211 for (
i = 0;
i <
s->nb_echoes;
i++) {
212 s->samples[
i] =
s->delay[
i] * outlink->sample_rate / 1000.0;
213 s->max_samples =
FFMAX(
s->max_samples,
s->samples[
i]);
214 volume +=
s->decay[
i];
217 if (
s->max_samples <= 0) {
221 s->fade_out =
s->max_samples;
223 if (volume *
s->in_gain *
s->out_gain > 1.0)
225 "out_gain %f can cause saturation of output\n",
s->out_gain);
227 switch (outlink->format) {
240 outlink->ch_layout.nb_channels,
267 if (
frame != out_frame)
277 int nb_samples =
FFMIN(
s->fade_out, 2048);
282 s->fade_out -= nb_samples;
289 s->echo_samples(
s,
s->delayptrs,
frame->extended_data,
frame->extended_data,
321 if (
s->eof &&
s->fade_out <= 0) {
344 .priv_class = &aecho_class,
static const AVFilterPad aecho_outputs[]
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
AVFILTER_DEFINE_CLASS(aecho)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
#define AVERROR_EOF
End of file.
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.
int av_samples_set_silence(uint8_t *const *audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
const char * name
Filter name.
int nb_channels
Number of channels in this layout.
A link between two filters.
const AVFilter ff_af_aecho
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
#define ECHO(name, type, min, max)
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
static av_cold int init(AVFilterContext *ctx)
static int config_output(AVFilterLink *outlink)
A filter pad used for either input or output.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
static av_cold void uninit(AVFilterContext *ctx)
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().
void(* echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs, uint8_t *const *src, uint8_t **dst, int nb_samples, int channels)
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
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)
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
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).
const AVFilterPad ff_audio_default_filterpad[1]
An AVFilterPad array whose only entry has name "default" and is of type AVMEDIA_TYPE_AUDIO.
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
static void fill_items(char *item_str, int *nb_items, float *items)
static const AVOption aecho_options[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
#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
FF_FILTER_FORWARD_WANTED(outlink, inlink)
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
int sample_rate
samples per second
#define i(width, name, range_min, range_max)
uint8_t ** extended_data
pointers to the data planes/channels.
const char * name
Pad name.
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)
static void count_items(char *item_str, int *nb_items)
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
@ AV_SAMPLE_FMT_DBLP
double, planar
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
static int activate(AVFilterContext *ctx)
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...
AVChannelLayout ch_layout
channel layout of current buffer (see libavutil/channel_layout.h)
#define FILTER_OUTPUTS(array)
#define FILTER_SAMPLEFMTS(...)