| @@ -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); | |||