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FFmpeg/libavfilter/af_silenceremove.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback 
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +02:00

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/*
* Copyright (c) 2001 Heikki Leinonen
* Copyright (c) 2001 Chris Bagwell
* Copyright (c) 2003 Donnie Smith
* Copyright (c) 2014 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <float.h> /* DBL_MAX */
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "libavutil/timestamp.h"
#include "audio.h"
#include "formats.h"
#include "avfilter.h"
#include "internal.h"
enum SilenceDetect {
D_PEAK,
D_RMS,
};
enum ThresholdMode {
T_ANY,
T_ALL,
};
enum SilenceMode {
SILENCE_TRIM,
SILENCE_TRIM_FLUSH,
SILENCE_COPY,
SILENCE_COPY_FLUSH,
SILENCE_STOP
};
typedef struct SilenceRemoveContext {
const AVClass *class;
enum SilenceMode mode;
int start_periods;
int64_t start_duration;
int64_t start_duration_opt;
double start_threshold;
int64_t start_silence;
int64_t start_silence_opt;
int start_mode;
int stop_periods;
int64_t stop_duration;
int64_t stop_duration_opt;
double stop_threshold;
int64_t stop_silence;
int64_t stop_silence_opt;
int stop_mode;
int64_t window_duration_opt;
AVFrame *start_holdoff;
AVFrame *start_silence_hold;
size_t start_holdoff_offset;
size_t start_holdoff_end;
size_t start_silence_offset;
size_t start_silence_end;
int start_found_periods;
AVFrame *stop_holdoff;
AVFrame *stop_silence_hold;
size_t stop_holdoff_offset;
size_t stop_holdoff_end;
size_t stop_silence_offset;
size_t stop_silence_end;
int stop_found_periods;
AVFrame *window;
int window_offset;
int64_t window_duration;
double sum;
int one_period;
int restart;
int64_t next_pts;
int detection;
void (*update)(struct SilenceRemoveContext *s, AVFrame *frame, int ch, int offset);
double (*compute)(struct SilenceRemoveContext *s, AVFrame *frame, int ch, int offset);
void (*copy)(struct SilenceRemoveContext *s, AVFrame *out, AVFrame *in,
int ch, int out_offset, int in_offset);
AVAudioFifo *fifo;
} SilenceRemoveContext;
#define OFFSET(x) offsetof(SilenceRemoveContext, x)
#define AF AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
static const AVOption silenceremove_options[] = {
{ "start_periods", "set periods of silence parts to skip from start", OFFSET(start_periods), AV_OPT_TYPE_INT, {.i64=0}, 0, 9000, AF },
{ "start_duration", "set start duration of non-silence part", OFFSET(start_duration_opt), AV_OPT_TYPE_DURATION, {.i64=0}, 0, INT32_MAX, AF },
{ "start_threshold", "set threshold for start silence detection", OFFSET(start_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, DBL_MAX, AF },
{ "start_silence", "set start duration of silence part to keep", OFFSET(start_silence_opt), AV_OPT_TYPE_DURATION, {.i64=0}, 0, INT32_MAX, AF },
{ "start_mode", "set which channel will trigger trimming from start", OFFSET(start_mode), AV_OPT_TYPE_INT, {.i64=T_ANY}, T_ANY, T_ALL, AF, "mode" },
{ "any", 0, 0, AV_OPT_TYPE_CONST, {.i64=T_ANY}, 0, 0, AF, "mode" },
{ "all", 0, 0, AV_OPT_TYPE_CONST, {.i64=T_ALL}, 0, 0, AF, "mode" },
{ "stop_periods", "set periods of silence parts to skip from end", OFFSET(stop_periods), AV_OPT_TYPE_INT, {.i64=0}, -9000, 9000, AF },
{ "stop_duration", "set stop duration of non-silence part", OFFSET(stop_duration_opt), AV_OPT_TYPE_DURATION, {.i64=0}, 0, INT32_MAX, AF },
{ "stop_threshold", "set threshold for stop silence detection", OFFSET(stop_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, DBL_MAX, AF },
{ "stop_silence", "set stop duration of silence part to keep", OFFSET(stop_silence_opt), AV_OPT_TYPE_DURATION, {.i64=0}, 0, INT32_MAX, AF },
{ "stop_mode", "set which channel will trigger trimming from end", OFFSET(stop_mode), AV_OPT_TYPE_INT, {.i64=T_ANY}, T_ANY, T_ALL, AF, "mode" },
{ "detection", "set how silence is detected", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=D_RMS}, D_PEAK,D_RMS, AF, "detection" },
{ "peak", "use absolute values of samples", 0, AV_OPT_TYPE_CONST, {.i64=D_PEAK},0, 0, AF, "detection" },
{ "rms", "use squared values of samples", 0, AV_OPT_TYPE_CONST, {.i64=D_RMS}, 0, 0, AF, "detection" },
{ "window", "set duration of window for silence detection", OFFSET(window_duration_opt), AV_OPT_TYPE_DURATION, {.i64=20000}, 0, 100000000, AF },
{ NULL }
};
AVFILTER_DEFINE_CLASS(silenceremove);
static void copy_double(SilenceRemoveContext *s, AVFrame *out, AVFrame *in,
int ch, int out_offset, int in_offset)
{
const double *srcp = (const double *)in->data[0];
const double src = srcp[in->channels * in_offset + ch];
double *dstp = (double *)out->data[0];
dstp[out->channels * out_offset + ch] = src;
}
static void copy_doublep(SilenceRemoveContext *s, AVFrame *out, AVFrame *in,
int ch, int out_offset, int in_offset)
{
const double *srcp = (const double *)in->extended_data[ch];
const double src = srcp[in_offset];
double *dstp = (double *)out->extended_data[ch];
dstp[out_offset] = src;
}
static void copy_float(SilenceRemoveContext *s, AVFrame *out, AVFrame *in,
int ch, int out_offset, int in_offset)
{
const float *srcp = (const float *)in->data[0];
const float src = srcp[in->channels * in_offset + ch];
float *dstp = (float *)out->data[0];
dstp[out->channels * out_offset + ch] = src;
}
static void copy_floatp(SilenceRemoveContext *s, AVFrame *out, AVFrame *in,
int ch, int out_offset, int in_offset)
{
const float *srcp = (const float *)in->extended_data[ch];
const float src = srcp[in_offset];
float *dstp = (float *)out->extended_data[ch];
dstp[out_offset] = src;
}
static double compute_peak_double(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->data[0];
const double *wsamples = (const double *)s->window->data[0];
double sample = samples[frame->channels * offset + ch];
double wsample = wsamples[frame->channels * s->window_offset + ch];
double new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmax(new_sum, 0.);
new_sum += fabs(sample);
return new_sum / s->window_duration;
}
static void update_peak_double(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->data[0];
double *wsamples = (double *)s->window->data[0];
double sample = samples[frame->channels * offset + ch];
double *wsample = &wsamples[frame->channels * s->window_offset + ch];
s->sum -= *wsample;
s->sum = fmax(s->sum, 0.);
*wsample = fabs(sample);
s->sum += *wsample;
}
static double compute_peak_float(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->data[0];
const float *wsamples = (const float *)s->window->data[0];
float sample = samples[frame->channels * offset + ch];
float wsample = wsamples[frame->channels * s->window_offset + ch];
float new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmaxf(new_sum, 0.f);
new_sum += fabsf(sample);
return new_sum / s->window_duration;
}
static void update_peak_float(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->data[0];
float *wsamples = (float *)s->window->data[0];
float sample = samples[frame->channels * offset + ch];
float *wsample = &wsamples[frame->channels * s->window_offset + ch];
s->sum -= *wsample;
s->sum = fmaxf(s->sum, 0.f);
*wsample = fabsf(sample);
s->sum += *wsample;
}
static double compute_rms_double(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->data[0];
const double *wsamples = (const double *)s->window->data[0];
double sample = samples[frame->channels * offset + ch];
double wsample = wsamples[frame->channels * s->window_offset + ch];
double new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmax(new_sum, 0.);
new_sum += sample * sample;
av_assert2(new_sum >= 0.);
return sqrt(new_sum / s->window_duration);
}
static void update_rms_double(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->data[0];
double *wsamples = (double *)s->window->data[0];
double sample = samples[frame->channels * offset + ch];
double *wsample = &wsamples[frame->channels * s->window_offset + ch];
s->sum -= *wsample;
s->sum = fmax(s->sum, 0.);
*wsample = sample * sample;
s->sum += *wsample;
}
static double compute_rms_float(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->data[0];
const float *wsamples = (const float *)s->window->data[0];
float sample = samples[frame->channels * offset + ch];
float wsample = wsamples[frame->channels * s->window_offset + ch];
float new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmaxf(new_sum, 0.f);
new_sum += sample * sample;
av_assert2(new_sum >= 0.f);
return sqrtf(new_sum / s->window_duration);
}
static void update_rms_float(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->data[0];
float sample = samples[frame->channels * offset + ch];
float *wsamples = (float *)s->window->data[0];
float *wsample = &wsamples[frame->channels * s->window_offset + ch];
s->sum -= *wsample;
s->sum = fmaxf(s->sum, 0.f);
*wsample = sample * sample;
s->sum += *wsample;
}
static double compute_peak_doublep(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->extended_data[ch];
const double *wsamples = (const double *)s->window->extended_data[ch];
double sample = samples[offset];
double wsample = wsamples[s->window_offset];
double new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmax(new_sum, 0.);
new_sum += fabs(sample);
return new_sum / s->window_duration;
}
static void update_peak_doublep(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->extended_data[ch];
double *wsamples = (double *)s->window->extended_data[ch];
double sample = samples[offset];
double *wsample = &wsamples[s->window_offset];
s->sum -= *wsample;
s->sum = fmax(s->sum, 0.);
*wsample = fabs(sample);
s->sum += *wsample;
}
static double compute_peak_floatp(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->extended_data[ch];
const float *wsamples = (const float *)s->window->extended_data[ch];
float sample = samples[offset];
float wsample = wsamples[s->window_offset];
float new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmaxf(new_sum, 0.f);
new_sum += fabsf(sample);
return new_sum / s->window_duration;
}
static void update_peak_floatp(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->extended_data[ch];
float *wsamples = (float *)s->window->extended_data[ch];
float sample = samples[offset];
float *wsample = &wsamples[s->window_offset];
s->sum -= *wsample;
s->sum = fmaxf(s->sum, 0.f);
*wsample = fabsf(sample);
s->sum += *wsample;
}
static double compute_rms_doublep(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->extended_data[ch];
const double *wsamples = (const double *)s->window->extended_data[ch];
double sample = samples[offset];
double wsample = wsamples[s->window_offset];
double new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmax(new_sum, 0.);
new_sum += sample * sample;
av_assert2(new_sum >= 0.);
return sqrt(new_sum / s->window_duration);
}
static void update_rms_doublep(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const double *samples = (const double *)frame->extended_data[ch];
double *wsamples = (double *)s->window->extended_data[ch];
double sample = samples[offset];
double *wsample = &wsamples[s->window_offset];
s->sum -= *wsample;
s->sum = fmax(s->sum, 0.);
*wsample = sample * sample;
s->sum += *wsample;
}
static double compute_rms_floatp(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->extended_data[ch];
const float *wsamples = (const float *)s->window->extended_data[ch];
float sample = samples[offset];
float wsample = wsamples[s->window_offset];
float new_sum;
new_sum = s->sum;
new_sum -= wsample;
new_sum = fmaxf(new_sum, 0.f);
new_sum += sample * sample;
av_assert2(new_sum >= 0.f);
return sqrtf(new_sum / s->window_duration);
}
static void update_rms_floatp(SilenceRemoveContext *s, AVFrame *frame, int ch, int offset)
{
const float *samples = (const float *)frame->extended_data[ch];
float *wsamples = (float *)s->window->extended_data[ch];
float sample = samples[offset];
float *wsample = &wsamples[s->window_offset];
s->sum -= *wsample;
s->sum = fmaxf(s->sum, 0.f);
*wsample = sample * sample;
s->sum += *wsample;
}
static av_cold int init(AVFilterContext *ctx)
{
SilenceRemoveContext *s = ctx->priv;
if (s->stop_periods < 0) {
s->stop_periods = -s->stop_periods;
s->restart = 1;
}
return 0;
}
static void clear_window(SilenceRemoveContext *s)
{
av_samples_set_silence(s->window->extended_data, 0, s->window_duration,
s->window->channels, s->window->format);
s->window_offset = 0;
s->sum = 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
SilenceRemoveContext *s = ctx->priv;
s->next_pts = AV_NOPTS_VALUE;
s->window_duration = av_rescale(s->window_duration_opt, inlink->sample_rate,
AV_TIME_BASE);
s->window_duration = FFMAX(1, s->window_duration);
s->window = ff_get_audio_buffer(ctx->outputs[0], s->window_duration);
if (!s->window)
return AVERROR(ENOMEM);
clear_window(s);
s->start_duration = av_rescale(s->start_duration_opt, inlink->sample_rate,
AV_TIME_BASE);
s->start_silence = av_rescale(s->start_silence_opt, inlink->sample_rate,
AV_TIME_BASE);
s->stop_duration = av_rescale(s->stop_duration_opt, inlink->sample_rate,
AV_TIME_BASE);
s->stop_silence = av_rescale(s->stop_silence_opt, inlink->sample_rate,
AV_TIME_BASE);
s->start_holdoff = ff_get_audio_buffer(ctx->outputs[0],
FFMAX(s->start_duration, 1));
if (!s->start_holdoff)
return AVERROR(ENOMEM);
s->start_silence_hold = ff_get_audio_buffer(ctx->outputs[0],
FFMAX(s->start_silence, 1));
if (!s->start_silence_hold)
return AVERROR(ENOMEM);
s->start_holdoff_offset = 0;
s->start_holdoff_end = 0;
s->start_found_periods = 0;
s->stop_holdoff = ff_get_audio_buffer(ctx->outputs[0],
FFMAX(s->stop_duration, 1));
if (!s->stop_holdoff)
return AVERROR(ENOMEM);
s->stop_silence_hold = ff_get_audio_buffer(ctx->outputs[0],
FFMAX(s->stop_silence, 1));
if (!s->stop_silence_hold)
return AVERROR(ENOMEM);
s->stop_holdoff_offset = 0;
s->stop_holdoff_end = 0;
s->stop_found_periods = 0;
if (s->start_periods)
s->mode = SILENCE_TRIM;
else
s->mode = SILENCE_COPY;
switch (inlink->format) {
case AV_SAMPLE_FMT_DBL:
s->copy = copy_double;
switch (s->detection) {
case D_PEAK:
s->update = update_peak_double;
s->compute = compute_peak_double;
break;
case D_RMS:
s->update = update_rms_double;
s->compute = compute_rms_double;
break;
}
break;
case AV_SAMPLE_FMT_FLT:
s->copy = copy_float;
switch (s->detection) {
case D_PEAK:
s->update = update_peak_float;
s->compute = compute_peak_float;
break;
case D_RMS:
s->update = update_rms_float;
s->compute = compute_rms_float;
break;
}
break;
case AV_SAMPLE_FMT_DBLP:
s->copy = copy_doublep;
switch (s->detection) {
case D_PEAK:
s->update = update_peak_doublep;
s->compute = compute_peak_doublep;
break;
case D_RMS:
s->update = update_rms_doublep;
s->compute = compute_rms_doublep;
break;
}
break;
case AV_SAMPLE_FMT_FLTP:
s->copy = copy_floatp;
switch (s->detection) {
case D_PEAK:
s->update = update_peak_floatp;
s->compute = compute_peak_floatp;
break;
case D_RMS:
s->update = update_rms_floatp;
s->compute = compute_rms_floatp;
break;
}
break;
default:
return AVERROR_BUG;
}
s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, 1024);
if (!s->fifo)
return AVERROR(ENOMEM);
return 0;
}
static void flush(SilenceRemoveContext *s,
AVFrame *out, AVFilterLink *outlink,
int *nb_samples_written, int flush_silence)
{
AVFrame *silence;
if (*nb_samples_written) {
out->nb_samples = *nb_samples_written;
av_audio_fifo_write(s->fifo, (void **)out->extended_data, out->nb_samples);
*nb_samples_written = 0;
}
av_frame_free(&out);
if (s->stop_silence_end <= 0 || !flush_silence)
return;
silence = ff_get_audio_buffer(outlink, s->stop_silence_end);
if (!silence)
return;
if (s->stop_silence_offset < s->stop_silence_end) {
av_samples_copy(silence->extended_data, s->stop_silence_hold->extended_data, 0,
s->stop_silence_offset,
s->stop_silence_end - s->stop_silence_offset,
outlink->channels, outlink->format);
}
if (s->stop_silence_offset > 0) {
av_samples_copy(silence->extended_data, s->stop_silence_hold->extended_data,
s->stop_silence_end - s->stop_silence_offset,
0, s->stop_silence_offset,
outlink->channels, outlink->format);
}
s->stop_silence_offset = 0;
s->stop_silence_end = 0;
av_audio_fifo_write(s->fifo, (void **)silence->extended_data, silence->nb_samples);
av_frame_free(&silence);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
SilenceRemoveContext *s = ctx->priv;
int nbs, nb_samples_read, nb_samples_written;
int i, j, threshold, ret = 0;
AVFrame *out;
nb_samples_read = nb_samples_written = 0;
if (s->next_pts == AV_NOPTS_VALUE)
s->next_pts = in->pts;
switch (s->mode) {
case SILENCE_TRIM:
silence_trim:
nbs = in->nb_samples - nb_samples_read;
if (!nbs)
break;
for (i = 0; i < nbs; i++) {
if (s->start_mode == T_ANY) {
threshold = 0;
for (j = 0; j < outlink->channels; j++) {
threshold |= s->compute(s, in, j, nb_samples_read) > s->start_threshold;
}
} else {
threshold = 1;
for (j = 0; j < outlink->channels; j++) {
threshold &= s->compute(s, in, j, nb_samples_read) > s->start_threshold;
}
}
if (threshold) {
for (j = 0; j < outlink->channels; j++) {
s->update(s, in, j, nb_samples_read);
s->copy(s, s->start_holdoff, in, j, s->start_holdoff_end, nb_samples_read);
}
s->window_offset++;
if (s->window_offset >= s->window_duration)
s->window_offset = 0;
s->start_holdoff_end++;
nb_samples_read++;
if (s->start_holdoff_end >= s->start_duration) {
s->start_found_periods += s->one_period >= 1;
s->one_period = 0;
if (s->start_found_periods >= s->start_periods) {
s->mode = SILENCE_TRIM_FLUSH;
goto silence_trim_flush;
}
s->start_holdoff_offset = 0;
s->start_holdoff_end = 0;
s->start_silence_offset = 0;
s->start_silence_end = 0;
}
} else {
s->start_holdoff_end = 0;
s->one_period++;
for (j = 0; j < outlink->channels; j++) {
s->update(s, in, j, nb_samples_read);
if (s->start_silence)
s->copy(s, s->start_silence_hold, in, j, s->start_silence_offset, nb_samples_read);
}
s->window_offset++;
if (s->window_offset >= s->window_duration)
s->window_offset = 0;
nb_samples_read++;
s->start_silence_offset++;
if (s->start_silence) {
s->start_silence_end = FFMIN(s->start_silence_end + 1, s->start_silence);
if (s->start_silence_offset >= s->start_silence)
s->start_silence_offset = 0;
}
}
}
break;
case SILENCE_TRIM_FLUSH:
silence_trim_flush:
nbs = s->start_holdoff_end - s->start_holdoff_offset;
if (!nbs)
break;
out = ff_get_audio_buffer(outlink, nbs + s->start_silence_end);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
if (s->start_silence_end > 0) {
if (s->start_silence_offset < s->start_silence_end) {
av_samples_copy(out->extended_data, s->start_silence_hold->extended_data, 0,
s->start_silence_offset,
s->start_silence_end - s->start_silence_offset,
outlink->channels, outlink->format);
}
if (s->start_silence_offset > 0) {
av_samples_copy(out->extended_data, s->start_silence_hold->extended_data,
s->start_silence_end - s->start_silence_offset,
0, s->start_silence_offset,
outlink->channels, outlink->format);
}
}
av_samples_copy(out->extended_data, s->start_holdoff->extended_data,
s->start_silence_end,
s->start_holdoff_offset, nbs,
outlink->channels, outlink->format);
s->start_holdoff_offset += nbs;
av_audio_fifo_write(s->fifo, (void **)out->extended_data, out->nb_samples);
av_frame_free(&out);
if (s->start_holdoff_offset == s->start_holdoff_end) {
s->start_holdoff_offset = 0;
s->start_holdoff_end = 0;
s->start_silence_offset = 0;
s->start_silence_end = 0;
s->mode = SILENCE_COPY;
goto silence_copy;
}
break;
case SILENCE_COPY:
silence_copy:
nbs = in->nb_samples - nb_samples_read;
if (!nbs)
break;
out = ff_get_audio_buffer(outlink, nbs);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
if (s->stop_periods) {
for (i = 0; i < nbs; i++) {
if (s->stop_mode == T_ANY) {
threshold = 0;
for (j = 0; j < outlink->channels; j++) {
threshold |= s->compute(s, in, j, nb_samples_read) > s->stop_threshold;
}
} else {
threshold = 1;
for (j = 0; j < outlink->channels; j++) {
threshold &= s->compute(s, in, j, nb_samples_read) > s->stop_threshold;
}
}
if (threshold && s->stop_holdoff_end && !s->stop_silence) {
s->mode = SILENCE_COPY_FLUSH;
flush(s, out, outlink, &nb_samples_written, 0);
s->one_period++;
goto silence_copy_flush;
} else if (threshold) {
for (j = 0; j < outlink->channels; j++) {
s->update(s, in, j, nb_samples_read);
s->copy(s, out, in, j, nb_samples_written, nb_samples_read);
}
s->window_offset++;
if (s->window_offset >= s->window_duration)
s->window_offset = 0;
nb_samples_read++;
nb_samples_written++;
s->one_period++;
} else if (!threshold) {
for (j = 0; j < outlink->channels; j++) {
s->update(s, in, j, nb_samples_read);
if (s->stop_silence)
s->copy(s, s->stop_silence_hold, in, j, s->stop_silence_offset, nb_samples_read);
s->copy(s, s->stop_holdoff, in, j, s->stop_holdoff_end, nb_samples_read);
}
if (s->stop_silence) {
s->stop_silence_offset++;
s->stop_silence_end = FFMIN(s->stop_silence_end + 1, s->stop_silence);
if (s->stop_silence_offset >= s->stop_silence) {
s->stop_silence_offset = 0;
}
}
s->window_offset++;
if (s->window_offset >= s->window_duration)
s->window_offset = 0;
nb_samples_read++;
s->stop_holdoff_end++;
if (s->stop_holdoff_end >= s->stop_duration) {
s->stop_found_periods += s->one_period >= 1;
s->one_period = 0;
if (s->stop_found_periods >= s->stop_periods) {
s->stop_holdoff_offset = 0;
s->stop_holdoff_end = 0;
if (!s->restart) {
s->mode = SILENCE_STOP;
flush(s, out, outlink, &nb_samples_written, 1);
goto silence_stop;
} else {
s->stop_found_periods = 0;
s->start_found_periods = 0;
s->start_holdoff_offset = 0;
s->start_holdoff_end = 0;
s->start_silence_offset = 0;
s->start_silence_end = 0;
clear_window(s);
s->mode = SILENCE_TRIM;
flush(s, out, outlink, &nb_samples_written, 1);
goto silence_trim;
}
}
s->mode = SILENCE_COPY_FLUSH;
flush(s, out, outlink, &nb_samples_written, 0);
goto silence_copy_flush;
}
}
}
s->one_period++;
flush(s, out, outlink, &nb_samples_written, 0);
} else {
av_samples_copy(out->extended_data, in->extended_data,
nb_samples_written,
nb_samples_read, nbs,
outlink->channels, outlink->format);
av_audio_fifo_write(s->fifo, (void **)out->extended_data, out->nb_samples);
av_frame_free(&out);
}
break;
case SILENCE_COPY_FLUSH:
silence_copy_flush:
nbs = s->stop_holdoff_end - s->stop_holdoff_offset;
if (!nbs)
break;
out = ff_get_audio_buffer(outlink, nbs);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_samples_copy(out->extended_data, s->stop_holdoff->extended_data, 0,
s->stop_holdoff_offset, nbs,
outlink->channels, outlink->format);
s->stop_holdoff_offset += nbs;
av_audio_fifo_write(s->fifo, (void **)out->extended_data, out->nb_samples);
av_frame_free(&out);
if (s->stop_holdoff_offset == s->stop_holdoff_end) {
s->stop_holdoff_offset = 0;
s->stop_holdoff_end = 0;
s->stop_silence_offset = 0;
s->stop_silence_end = 0;
s->mode = SILENCE_COPY;
goto silence_copy;
}
break;
case SILENCE_STOP:
silence_stop:
break;
default:
ret = AVERROR_BUG;
}
av_frame_free(&in);
if (av_audio_fifo_size(s->fifo) > 0) {
out = ff_get_audio_buffer(outlink, av_audio_fifo_size(s->fifo));
if (!out)
return AVERROR(ENOMEM);
av_audio_fifo_read(s->fifo, (void **)out->extended_data, out->nb_samples);
out->pts = s->next_pts;
s->next_pts += av_rescale_q(out->nb_samples,
(AVRational){1, outlink->sample_rate},
outlink->time_base);
ret = ff_filter_frame(outlink, out);
}
return ret;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
SilenceRemoveContext *s = ctx->priv;
int ret;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF && (s->mode == SILENCE_COPY_FLUSH ||
s->mode == SILENCE_COPY)) {
int nbs = s->stop_holdoff_end - s->stop_holdoff_offset;
if (nbs) {
AVFrame *frame;
frame = ff_get_audio_buffer(outlink, nbs);
if (!frame)
return AVERROR(ENOMEM);
av_samples_copy(frame->extended_data, s->stop_holdoff->extended_data, 0,
s->stop_holdoff_offset, nbs,
outlink->channels, outlink->format);
frame->pts = s->next_pts;
s->next_pts += av_rescale_q(frame->nb_samples,
(AVRational){1, outlink->sample_rate},
outlink->time_base);
ret = ff_filter_frame(outlink, frame);
}
s->mode = SILENCE_STOP;
}
return ret;
}
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static av_cold void uninit(AVFilterContext *ctx)
{
SilenceRemoveContext *s = ctx->priv;
av_frame_free(&s->start_holdoff);
av_frame_free(&s->start_silence_hold);
av_frame_free(&s->stop_holdoff);
av_frame_free(&s->stop_silence_hold);
av_frame_free(&s->window);
av_audio_fifo_free(s->fifo);
s->fifo = NULL;
}
static const AVFilterPad silenceremove_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
static const AVFilterPad silenceremove_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
},
};
const AVFilter ff_af_silenceremove = {
.name = "silenceremove",
.description = NULL_IF_CONFIG_SMALL("Remove silence."),
.priv_size = sizeof(SilenceRemoveContext),
.priv_class = &silenceremove_class,
.init = init,
.uninit = uninit,
FILTER_INPUTS(silenceremove_inputs),
FILTER_OUTPUTS(silenceremove_outputs),
FILTER_QUERY_FUNC(query_formats),
};
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