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ffmpeg.wasm-core
Commit a8536236 authored by Paul B Mahol's avatar Paul B Mahol
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avfilter/af_silenceremove: prefer outlink instead of inlink

parent 346b2323
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Showing with 28 additions and 28 deletions
libavfilter/af_silenceremove.c
View file @ a8536236
......@@ -321,31 +321,31 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
switch (s->mode) {
case SILENCE_TRIM:
silence_trim:
nbs = in->nb_samples - nb_samples_read / inlink->channels;
nbs = in->nb_samples - nb_samples_read / outlink->channels;
if (!nbs)
break;
for (i = 0; i < nbs; i++) {
if (s->start_mode) {
threshold = 0;
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
threshold |= s->compute(s, ibuf[j]) > s->start_threshold;
}
} else {
threshold = 1;
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
threshold &= s->compute(s, ibuf[j]) > s->start_threshold;
}
}
if (threshold) {
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
s->update(s, *ibuf);
s->start_holdoff[s->start_holdoff_end++] = *ibuf++;
}
nb_samples_read += inlink->channels;
nb_samples_read += outlink->channels;
if (s->start_holdoff_end >= s->start_duration * inlink->channels) {
if (s->start_holdoff_end >= s->start_duration * outlink->channels) {
if (++s->start_found_periods >= s->start_periods) {
s->mode = SILENCE_TRIM_FLUSH;
goto silence_trim_flush;
......@@ -359,19 +359,19 @@ silence_trim:
} else {
s->start_holdoff_end = 0;
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
s->update(s, ibuf[j]);
if (s->start_silence) {
s->start_silence_hold[s->start_silence_offset++] = ibuf[j];
s->start_silence_end = FFMIN(s->start_silence_end + 1, inlink->channels * s->start_silence);
if (s->start_silence_offset >= inlink->channels * s->start_silence) {
s->start_silence_end = FFMIN(s->start_silence_end + 1, outlink->channels * s->start_silence);
if (s->start_silence_offset >= outlink->channels * s->start_silence) {
s->start_silence_offset = 0;
}
}
}
ibuf += inlink->channels;
nb_samples_read += inlink->channels;
ibuf += outlink->channels;
nb_samples_read += outlink->channels;
}
}
break;
......@@ -379,11 +379,11 @@ silence_trim:
case SILENCE_TRIM_FLUSH:
silence_trim_flush:
nbs = s->start_holdoff_end - s->start_holdoff_offset;
nbs -= nbs % inlink->channels;
nbs -= nbs % outlink->channels;
if (!nbs)
break;
out = ff_get_audio_buffer(inlink, nbs / inlink->channels + s->start_silence_end / inlink->channels);
out = ff_get_audio_buffer(outlink, nbs / outlink->channels + s->start_silence_end / outlink->channels);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
......@@ -428,11 +428,11 @@ silence_trim_flush:
case SILENCE_COPY:
silence_copy:
nbs = in->nb_samples - nb_samples_read / inlink->channels;
nbs = in->nb_samples - nb_samples_read / outlink->channels;
if (!nbs)
break;
out = ff_get_audio_buffer(inlink, nbs);
out = ff_get_audio_buffer(outlink, nbs);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
......@@ -443,12 +443,12 @@ silence_copy:
for (i = 0; i < nbs; i++) {
if (s->stop_mode) {
threshold = 0;
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
threshold |= s->compute(s, ibuf[j]) > s->stop_threshold;
}
} else {
threshold = 1;
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
threshold &= s->compute(s, ibuf[j]) > s->stop_threshold;
}
}
......@@ -458,28 +458,28 @@ silence_copy:
flush(s, out, outlink, &nb_samples_written, &ret, 0);
goto silence_copy_flush;
} else if (threshold) {
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
s->update(s, *ibuf);
*obuf++ = *ibuf++;
}
nb_samples_read += inlink->channels;
nb_samples_written += inlink->channels;
nb_samples_read += outlink->channels;
nb_samples_written += outlink->channels;
} else if (!threshold) {
for (j = 0; j < inlink->channels; j++) {
for (j = 0; j < outlink->channels; j++) {
s->update(s, *ibuf);
if (s->stop_silence) {
s->stop_silence_hold[s->stop_silence_offset++] = *ibuf;
s->stop_silence_end = FFMIN(s->stop_silence_end + 1, inlink->channels * s->stop_silence);
if (s->stop_silence_offset >= inlink->channels * s->stop_silence) {
s->stop_silence_end = FFMIN(s->stop_silence_end + 1, outlink->channels * s->stop_silence);
if (s->stop_silence_offset >= outlink->channels * s->stop_silence) {
s->stop_silence_offset = 0;
}
}
s->stop_holdoff[s->stop_holdoff_end++] = *ibuf++;
}
nb_samples_read += inlink->channels;
nb_samples_read += outlink->channels;
if (s->stop_holdoff_end >= s->stop_duration * inlink->channels) {
if (s->stop_holdoff_end >= s->stop_duration * outlink->channels) {
if (++s->stop_found_periods >= s->stop_periods) {
s->stop_holdoff_offset = 0;
s->stop_holdoff_end = 0;
......@@ -509,7 +509,7 @@ silence_copy:
}
flush(s, out, outlink, &nb_samples_written, &ret, 0);
} else {
memcpy(obuf, ibuf, sizeof(double) * nbs * inlink->channels);
memcpy(obuf, ibuf, sizeof(double) * nbs * outlink->channels);
out->pts = s->next_pts;
s->next_pts += av_rescale_q(out->nb_samples,
......@@ -523,11 +523,11 @@ silence_copy:
case SILENCE_COPY_FLUSH:
silence_copy_flush:
nbs = s->stop_holdoff_end - s->stop_holdoff_offset;
nbs -= nbs % inlink->channels;
nbs -= nbs % outlink->channels;
if (!nbs)
break;
out = ff_get_audio_buffer(inlink, nbs / inlink->channels);
out = ff_get_audio_buffer(outlink, nbs / outlink->channels);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
......
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