Go to the documentation of this file.
48 #define OFFSET(x) offsetof(AudioEchoContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
66 for (p = item_str; *p; p++) {
73 static void fill_items(
char *item_str,
int *nb_items,
float *items)
75 char *p, *saveptr =
NULL;
76 int i, new_nb_items = 0;
79 for (
i = 0;
i < *nb_items;
i++) {
83 new_nb_items +=
av_sscanf(tstr,
"%f", &items[new_nb_items]) == 1;
86 *nb_items = new_nb_items;
105 int nb_delays, nb_decays,
i;
107 if (!
s->delays || !
s->decays) {
117 if (!
s->delay || !
s->decay)
123 if (nb_delays != nb_decays) {
124 av_log(
ctx,
AV_LOG_ERROR,
"Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
128 s->nb_echoes = nb_delays;
138 for (
i = 0;
i < nb_delays;
i++) {
139 if (
s->delay[
i] <= 0 ||
s->delay[
i] > 90000) {
143 if (
s->decay[
i] <= 0 ||
s->decay[
i] > 1) {
186 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
188 #define ECHO(name, type, min, max) \
189 static void echo_samples_## name ##p(AudioEchoContext *ctx, \
190 uint8_t **delayptrs, \
191 uint8_t * const *src, uint8_t **dst, \
192 int nb_samples, int channels) \
194 const double out_gain = ctx->out_gain; \
195 const double in_gain = ctx->in_gain; \
196 const int nb_echoes = ctx->nb_echoes; \
197 const int max_samples = ctx->max_samples; \
198 int i, j, chan, av_uninit(index); \
200 av_assert1(channels > 0); \
202 for (chan = 0; chan < channels; chan++) { \
203 const type *s = (type *)src[chan]; \
204 type *d = (type *)dst[chan]; \
205 type *dbuf = (type *)delayptrs[chan]; \
207 index = ctx->delay_index; \
208 for (i = 0; i < nb_samples; i++, s++, d++) { \
212 out = in * in_gain; \
213 for (j = 0; j < nb_echoes; j++) { \
214 int ix = index + max_samples - ctx->samples[j]; \
215 ix = MOD(ix, max_samples); \
216 out += dbuf[ix] * ctx->decay[j]; \
220 *d = av_clipd(out, min, max); \
223 index = MOD(index + 1, max_samples); \
226 ctx->delay_index = index; \
229 ECHO(dbl,
double, -1.0, 1.0 )
230 ECHO(flt,
float, -1.0, 1.0 )
231 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
241 for (
i = 0;
i <
s->nb_echoes;
i++) {
242 s->samples[
i] =
s->delay[
i] * outlink->sample_rate / 1000.0;
243 s->max_samples =
FFMAX(
s->max_samples,
s->samples[
i]);
244 volume +=
s->decay[
i];
247 if (
s->max_samples <= 0) {
251 s->fade_out =
s->max_samples;
253 if (volume *
s->in_gain *
s->out_gain > 1.0)
255 "out_gain %f can cause saturation of output\n",
s->out_gain);
257 switch (outlink->format) {
297 if (
frame != out_frame)
307 int nb_samples =
FFMIN(
s->fade_out, 2048);
312 s->fade_out -= nb_samples;
319 s->echo_samples(
s,
s->delayptrs,
frame->extended_data,
frame->extended_data,
351 if (
s->eof &&
s->fade_out <= 0) {
384 .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
A list of supported channel layouts.
#define AV_LOG_WARNING
Something somehow does not look correct.
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
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)
static int query_formats(AVFilterContext *ctx)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static enum AVSampleFormat sample_fmts[]
enum MovChannelLayoutTag * layouts
#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.
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
const char * name
Filter name.
A link between two filters.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
int channels
Number of channels.
#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)
static const AVFilterPad outputs[]
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static const AVFilterPad aecho_inputs[]
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
#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).
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 inputs
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
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 sample_rate
samples per second
#define i(width, name, range_min, range_max)
uint8_t ** extended_data
pointers to the data planes/channels.
AVSampleFormat
Audio sample formats.
const char * name
Pad name.
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)
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...