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authorMichael Niedermayer <michaelni@gmx.at>2012-05-19 20:48:53 +0400
committerMichael Niedermayer <michaelni@gmx.at>2012-05-19 21:23:38 +0400
commit21d8a80e30e9e2050dbc2335670028331d6dff95 (patch)
tree4529b5c973c3a53a67703096187a95ac1a20c868 /libavfilter/af_aresample.c
parent087d09b6d52285b88ddb6199d3386496e5403684 (diff)
af_aresample: use new swr API to pass and compensate PTS
This code is not only much more powerfull its also simpler Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavfilter/af_aresample.c')
-rw-r--r--libavfilter/af_aresample.c17
1 files changed, 16 insertions, 1 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
index cc479e802a..3b9fe9d617 100644
--- a/libavfilter/af_aresample.c
+++ b/libavfilter/af_aresample.c
@@ -182,6 +182,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
+#if 0
if(insamplesref->pts != AV_NOPTS_VALUE) {
aresample->next_pts =
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)
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);
avfilter_unref_buffer(insamplesref);
}
@@ -201,6 +211,7 @@ static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AResampleContext *aresample = ctx->priv;
+ AVFilterLink *const inlink = outlink->src->inputs[0];
int ret = avfilter_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF) {
@@ -218,9 +229,13 @@ static int request_frame(AVFilterLink *outlink)
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
+#if 0
outsamplesref->pts = aresample->next_pts;
if(aresample->next_pts != AV_NOPTS_VALUE)
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);
return 0;