static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref) { AResampleContext *aresample = inlink->dst->priv; const int n_in = insamplesref->nb_samples; int64_t delay; int n_out = n_in * aresample->ratio + 32; AVFilterLink *const outlink = inlink->dst->outputs[0]; AVFrame *outsamplesref; int ret; delay = swr_get_delay(aresample->swr, outlink->sample_rate); if (delay > 0) n_out += FFMIN(delay, FFMAX(4096, n_out)); outsamplesref = ff_get_audio_buffer(outlink, n_out); if(!outsamplesref) return AVERROR(ENOMEM); av_frame_copy_props(outsamplesref, insamplesref); outsamplesref->format = outlink->format; av_frame_set_channels(outsamplesref, outlink->channels); outsamplesref->channel_layout = outlink->channel_layout; outsamplesref->sample_rate = outlink->sample_rate; if(insamplesref->pts != AV_NOPTS_VALUE) { int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den); int64_t outpts= swr_next_pts(aresample->swr, inpts); aresample->next_pts = outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate); } else { outsamplesref->pts = AV_NOPTS_VALUE; } n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, (void *)insamplesref->extended_data, n_in); if (n_out <= 0) { av_frame_free(&outsamplesref); av_frame_free(&insamplesref); return 0; } aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers outsamplesref->nb_samples = n_out; ret = ff_filter_frame(outlink, outsamplesref); aresample->req_fullfilled= 1; av_frame_free(&insamplesref); return ret; }