diff --git a/ffmpeg.c b/ffmpeg.c
old mode 100644
new mode 100755
index acaa523..5cb498d
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -1521,6 +1521,12 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
                decoded_frame->sample_rate, av_get_sample_fmt_name(decoded_frame->format),
                avctx->channels, layout2);
 
+
+        int prev_samp_rate,prev_chan_layout;
+
+        prev_samp_rate = ist->resample_sample_rate;
+        prev_chan_layout = ist->resample_channel_layout;
+
         ist->resample_sample_fmt     = decoded_frame->format;
         ist->resample_sample_rate    = decoded_frame->sample_rate;
         ist->resample_channel_layout = decoded_frame->channel_layout;
@@ -1534,6 +1540,10 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
                     av_log(NULL, AV_LOG_FATAL, "Error reinitializing filters!\n");
                     exit(1);
                 }
+
+                for (int i_sr = 0; i_sr < ist->nb_filters; i_sr++)
+                    update_prev_sr_chs(ist->filters[i_sr]->filter,prev_samp_rate,prev_chan_layout);
+
                 for (j = 0; j < fg->nb_outputs; j++) {
                     OutputStream *ost = fg->outputs[j]->ost;
                     if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
old mode 100644
new mode 100755
index 17b7630..ae13284
--- a/libavfilter/af_aresample.c
+++ b/libavfilter/af_aresample.c
@@ -190,6 +190,16 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
 
     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);
+
+        if((inlink->prev_sample_rate && inlink->prev_sample_rate != inlink->sample_rate) || (inlink->prev_channel_layout && inlink->channel_layout != inlink->prev_channel_layout))
+        {
+            av_log(NULL, AV_LOG_INFO, "change in ingest samplerate or channels sr:%d,prev_sr:%d,ch:%d,prev_ch:%d\n", inlink->sample_rate,inlink->prev_sample_rate,inlink->channel_layout,inlink->prev_channel_layout);
+            // set flag to recalculate the aresample->swr->outpts which becomes null at this point.
+            swr_set_recalc_outpts(aresample->swr);
+        }
+        inlink->prev_sample_rate = inlink->sample_rate;
+        inlink->prev_channel_layout = inlink->channel_layout;
+
         int64_t outpts= swr_next_pts(aresample->swr, inpts);
         aresample->next_pts =
         outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
diff --git a/libavfilter/avcodec.h b/libavfilter/avcodec.h
old mode 100644
new mode 100755
index 5f4209a..1f18c86
--- a/libavfilter/avcodec.h
+++ b/libavfilter/avcodec.h
@@ -128,4 +128,7 @@ int avfilter_fill_frame_from_buffer_ref(AVFrame *frame,
 int av_buffersrc_add_frame(AVFilterContext *buffer_src,
                            const AVFrame *frame, int flags);
 
+void update_prev_sr_chs(AVFilterContext *buffer_src,int prev_sr,int prev_ch);
+
+
 #endif /* AVFILTER_AVCODEC_H */
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
old mode 100644
new mode 100755
index 1c80167..589a461
--- a/libavfilter/avfilter.h
+++ b/libavfilter/avfilter.h
@@ -530,6 +530,10 @@ struct AVFilterLink {
     uint64_t channel_layout;    ///< channel layout of current buffer (see libavutil/channel_layout.h)
     int sample_rate;            ///< samples per second
 
+
+    uint64_t prev_channel_layout;
+    int prev_sample_rate;
+
     int format;                 ///< agreed upon media format
 
     /**
diff --git a/libavfilter/buffersrc.c b/libavfilter/buffersrc.c
old mode 100644
new mode 100755
index 5d0fa92..8003d6f
--- a/libavfilter/buffersrc.c
+++ b/libavfilter/buffersrc.c
@@ -73,6 +73,12 @@ typedef struct {
         return AVERROR(EINVAL);\
     }
 
+void update_prev_sr_chs(AVFilterContext *buffer_src,int prev_samp_rate,int prev_chan_layout)
+{
+    buffer_src->outputs[0]->prev_sample_rate = prev_samp_rate;
+    buffer_src->outputs[0]->prev_channel_layout = prev_chan_layout;
+}
+
 int av_buffersrc_add_frame(AVFilterContext *buffer_src,
                            const AVFrame *frame, int flags)
 {
diff --git a/libswresample/swresample.c b/libswresample/swresample.c
old mode 100644
new mode 100755
index f01927f..6700e33
--- a/libswresample/swresample.c
+++ b/libswresample/swresample.c
@@ -226,6 +226,9 @@ av_cold int swr_init(struct SwrContext *s){
     s->in_buffer_index= 0;
     s->in_buffer_count= 0;
     s->resample_in_constraint= 0;
+
+    s->flag_to_recalc_outpts = 0;
+
     free_temp(&s->postin);
     free_temp(&s->midbuf);
     free_temp(&s->preout);
@@ -823,7 +826,9 @@ int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensatio
 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
     if(pts == INT64_MIN)
         return s->outpts;
-    if(s->min_compensation >= FLT_MAX) {
+    
+    if(s->min_compensation >= FLT_MAX || (!s->outpts && s->flag_to_recalc_outpts)) {
+//    if(s->min_compensation >= FLT_MAX) {
         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
     } else {
         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
@@ -849,3 +854,6 @@ int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
         return s->outpts;
     }
 }
+void swr_set_recalc_outpts(struct SwrContext *s){
+    s->flag_to_recalc_outpts = 1;
+}
diff --git a/libswresample/swresample.h b/libswresample/swresample.h
old mode 100644
new mode 100755
index 1c6090d..ae703ea
--- a/libswresample/swresample.h
+++ b/libswresample/swresample.h
@@ -222,6 +222,7 @@ int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
  */
 int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
 
+void swr_set_recalc_outpts(struct SwrContext *s);
 /**
  * Activate resampling compensation.
  */
diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h
old mode 100644
new mode 100755
index 70a361b..7853c95
--- a/libswresample/swresample_internal.h
+++ b/libswresample/swresample_internal.h
@@ -102,6 +102,7 @@ struct SwrContext {
     int resample_in_constraint;                     ///< 1 if the input end was reach before the output end, 0 otherwise
     int flushed;                                    ///< 1 if data is to be flushed and no further input is expected
     int64_t outpts;                                 ///< output PTS
+    int flag_to_recalc_outpts;
     int drop_output;                                ///< number of output samples to drop
 
     struct AudioConvert *in_convert;                ///< input conversion context