80 #define OFFSET(x) offsetof(SilenceRemoveContext, x) 81 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM 105 new_sum += fabs(sample);
127 new_sum += sample *
sample;
231 int *nb_samples_written,
int *ret)
233 if (*nb_samples_written) {
242 *nb_samples_written = 0;
253 int i, j, threshold, ret = 0;
254 int nbs, nb_samples_read, nb_samples_written;
255 double *obuf, *ibuf = (
double *)in->
data[0];
258 nb_samples_read = nb_samples_written = 0;
267 for (i = 0; i < nbs; i++) {
269 for (j = 0; j < inlink->
channels; j++) {
274 for (j = 0; j < inlink->
channels; j++) {
278 nb_samples_read += inlink->
channels;
283 goto silence_trim_flush;
292 for (j = 0; j < inlink->
channels; j++)
296 nb_samples_read += inlink->
channels;
315 nbs *
sizeof(
double));
345 obuf = (
double *)
out->data[0];
348 for (i = 0; i < nbs; i++) {
350 for (j = 0; j < inlink->
channels; j++)
355 flush(s,
out, outlink, &nb_samples_written, &ret);
356 goto silence_copy_flush;
357 }
else if (threshold) {
358 for (j = 0; j < inlink->
channels; j++) {
362 nb_samples_read += inlink->
channels;
363 nb_samples_written += inlink->
channels;
364 }
else if (!threshold) {
365 for (j = 0; j < inlink->
channels; j++) {
369 nb_samples_written++;
374 nb_samples_read += inlink->
channels;
383 flush(s,
out, outlink, &nb_samples_written, &ret);
392 flush(s,
out, outlink, &nb_samples_written, &ret);
397 flush(s,
out, outlink, &nb_samples_written, &ret);
398 goto silence_copy_flush;
402 flush(s,
out, outlink, &nb_samples_written, &ret);
404 memcpy(obuf, ibuf,
sizeof(
double) * nbs * inlink->
channels);
429 nbs *
sizeof(
double));
474 nbs *
sizeof(
double));
546 .
name =
"silenceremove",
549 .priv_class = &silenceremove_class,
553 .
inputs = silenceremove_inputs,
554 .
outputs = silenceremove_outputs,
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
void(* update)(struct SilenceRemoveContext *s, double sample)
This structure describes decoded (raw) audio or video data.
static const AVFilterPad silenceremove_outputs[]
#define AV_LOG_WARNING
Something somehow does not look correct.
Main libavfilter public API header.
AVFilter ff_af_silenceremove
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).
static av_cold int init(AVFilterContext *ctx)
#define AVERROR_EOF
End of file.
A filter pad used for either input or output.
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...
double(* compute)(struct SilenceRemoveContext *s, double sample)
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[]
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 void flush(SilenceRemoveContext *s, AVFrame *out, AVFilterLink *outlink, int *nb_samples_written, int *ret)
static double compute_peak(SilenceRemoveContext *s, double sample)
AVSampleFormat
Audio sample formats.
typedef void(RENAME(mix_any_func_type))
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_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)
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)
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