103 #define OFFSET(x) offsetof(SilenceRemoveContext, x)
104 #define AF AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
134 new_sum += fabs(sample);
156 new_sum += sample *
sample;
267 int *nb_samples_written,
int *ret,
int flush_silence)
271 if (*nb_samples_written) {
282 *nb_samples_written = 0;
287 if (
s->stop_silence_end <= 0 || !flush_silence)
296 if (
s->stop_silence_offset <
s->stop_silence_end) {
297 memcpy(silence->data[0],
298 &
s->stop_silence_hold[
s->stop_silence_offset],
299 (
s->stop_silence_end -
s->stop_silence_offset) *
sizeof(
double));
302 if (
s->stop_silence_offset > 0) {
303 memcpy(silence->data[0] + (
s->stop_silence_end -
s->stop_silence_offset) *
sizeof(
double),
304 &
s->stop_silence_hold[0],
305 s->stop_silence_offset *
sizeof(
double));
308 s->stop_silence_offset = 0;
309 s->stop_silence_end = 0;
311 silence->pts =
s->next_pts;
324 int i, j, threshold, ret = 0;
325 int nbs, nb_samples_read, nb_samples_written;
326 double *obuf, *ibuf = (
double *)in->
data[0];
329 nb_samples_read = nb_samples_written = 0;
338 for (i = 0; i < nbs; i++) {
341 for (j = 0; j < outlink->
channels; j++) {
346 for (j = 0; j < outlink->
channels; j++) {
352 for (j = 0; j < outlink->
channels; j++) {
356 nb_samples_read += outlink->
channels;
361 goto silence_trim_flush;
372 for (j = 0; j < outlink->
channels; j++) {
384 nb_samples_read += outlink->
channels;
418 nbs *
sizeof(
double));
450 obuf = (
double *)
out->data[0];
453 for (i = 0; i < nbs; i++) {
456 for (j = 0; j < outlink->
channels; j++) {
461 for (j = 0; j < outlink->
channels; j++) {
468 flush(s,
out, outlink, &nb_samples_written, &ret, 0);
469 goto silence_copy_flush;
470 }
else if (threshold) {
471 for (j = 0; j < outlink->
channels; j++) {
475 nb_samples_read += outlink->
channels;
476 nb_samples_written += outlink->
channels;
477 }
else if (!threshold) {
478 for (j = 0; j < outlink->
channels; j++) {
490 nb_samples_read += outlink->
channels;
499 flush(s,
out, outlink, &nb_samples_written, &ret, 1);
510 flush(s,
out, outlink, &nb_samples_written, &ret, 1);
515 flush(s,
out, outlink, &nb_samples_written, &ret, 0);
516 goto silence_copy_flush;
520 flush(s,
out, outlink, &nb_samples_written, &ret, 0);
522 memcpy(obuf, ibuf,
sizeof(
double) * nbs * outlink->
channels);
535 nbs =
s->stop_holdoff_end -
s->stop_holdoff_offset;
536 nbs -= nbs % outlink->channels;
546 memcpy(
out->data[0], &
s->stop_holdoff[
s->stop_holdoff_offset],
547 nbs *
sizeof(
double));
548 s->stop_holdoff_offset += nbs;
550 out->pts =
s->next_pts;
557 if (
s->stop_holdoff_offset ==
s->stop_holdoff_end) {
558 s->stop_holdoff_offset = 0;
559 s->stop_holdoff_end = 0;
560 s->stop_silence_offset = 0;
561 s->stop_silence_end = 0;
594 nbs *
sizeof(
double));
668 .
name =
"silenceremove",
671 .priv_class = &silenceremove_class,
675 .
inputs = silenceremove_inputs,
676 .
outputs = silenceremove_outputs,
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
This structure describes decoded (raw) audio or video data.
static void flush(SilenceRemoveContext *s, AVFrame *out, AVFilterLink *outlink, int *nb_samples_written, int *ret, int flush_silence)
double * start_silence_hold
static const AVFilterPad silenceremove_outputs[]
Main libavfilter public API header.
void(* update)(struct SilenceRemoveContext *s, double sample)
AVFilter ff_af_silenceremove
int64_t start_silence_opt
double(* compute)(struct SilenceRemoveContext *s, double sample)
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
timestamp utils, mostly useful for debugging/logging purposes
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
size_t stop_silence_offset
static av_cold int init(AVFilterContext *ctx)
#define AVERROR_EOF
End of file.
A filter pad used for either input or output.
int64_t start_duration_opt
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
A link between two filters.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void * priv
private data for use by the filter
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
#define AV_TIME_BASE
Internal time base represented as integer.
static const AVFilterPad silenceremove_inputs[]
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
static int request_frame(AVFilterLink *outlink)
AVFILTER_DEFINE_CLASS(silenceremove)
AVFilterContext * src
source filter
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
A list of supported channel layouts.
static double compute_peak(SilenceRemoveContext *s, double sample)
AVSampleFormat
Audio sample formats.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
static av_cold void uninit(AVFilterContext *ctx)
Describe the class of an AVClass context structure.
static void update_rms(SilenceRemoveContext *s, double sample)
size_t start_silence_offset
Rational number (pair of numerator and denominator).
static double compute_rms(SilenceRemoveContext *s, double sample)
static void clear_window(SilenceRemoveContext *s)
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
size_t stop_holdoff_offset
static void update_peak(SilenceRemoveContext *s, double sample)
int64_t stop_duration_opt
int channels
Number of channels.
static const AVOption silenceremove_options[]
AVFilterContext * dst
dest filter
size_t start_holdoff_offset
static enum AVSampleFormat sample_fmts[]
#define av_malloc_array(a, b)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
int nb_samples
number of audio samples (per channel) described by this frame
double * stop_silence_hold