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af_aresample: use new swr API to pass and compensate PTS

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This code is not only much more powerfull its also simpler

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-05-19 18:48:53 +02:00
parent 087d09b6d5
commit 21d8a80e30
1 changed files with 16 additions and 1 deletions
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17
libavfilter/af_aresample.c
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@@ -182,6 +182,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
outsamplesref->audio->sample_rate = outlink->sample_rate; outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out; outsamplesref->audio->nb_samples = n_out;
#if 0
if(insamplesref->pts != AV_NOPTS_VALUE) { if(insamplesref->pts != AV_NOPTS_VALUE) {
aresample->next_pts = aresample->next_pts =
outsamplesref->pts = av_rescale_q(insamplesref->pts, inlink->time_base, outlink->time_base) outsamplesref->pts = av_rescale_q(insamplesref->pts, inlink->time_base, outlink->time_base)
@@ -192,7 +193,16 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
} }
if(aresample->next_pts != AV_NOPTS_VALUE) if(aresample->next_pts != AV_NOPTS_VALUE)
aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base); aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
#else
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 = (outpts + inlink->sample_rate/2) / inlink->sample_rate;
} else {
outsamplesref->pts = AV_NOPTS_VALUE;
}
#endif
ff_filter_samples(outlink, outsamplesref); ff_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref); avfilter_unref_buffer(insamplesref);
} }
@@ -201,6 +211,7 @@ static int request_frame(AVFilterLink *outlink)
{ {
AVFilterContext *ctx = outlink->src; AVFilterContext *ctx = outlink->src;
AResampleContext *aresample = ctx->priv; AResampleContext *aresample = ctx->priv;
AVFilterLink *const inlink = outlink->src->inputs[0];
int ret = avfilter_request_frame(ctx->inputs[0]); int ret = avfilter_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF) { if (ret == AVERROR_EOF) {
@@ -218,9 +229,13 @@ static int request_frame(AVFilterLink *outlink)
outsamplesref->audio->sample_rate = outlink->sample_rate; outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out; outsamplesref->audio->nb_samples = n_out;
#if 0
outsamplesref->pts = aresample->next_pts; outsamplesref->pts = aresample->next_pts;
if(aresample->next_pts != AV_NOPTS_VALUE) if(aresample->next_pts != AV_NOPTS_VALUE)
aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base); aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
#else
outsamplesref->pts = (swr_next_pts(aresample->swr, INT64_MIN) + inlink->sample_rate/2) / inlink->sample_rate;
#endif
ff_filter_samples(outlink, outsamplesref); ff_filter_samples(outlink, outsamplesref);
return 0; return 0;