[FFmpeg-cvslog] avfilter/af_atempo: Make ffplay display correct timestamps when seeking

[Pavel Koshevoy]

ffmpeg | branch: master | Pavel Koshevoy <pkoshevoy at gmail.com> | Tue May  7 22:39:45 2019 -0600| [181031906e4984c1c539dc84d21386a4397e257e] | committer: Pavel Koshevoy

avfilter/af_atempo: Make ffplay display correct timestamps when seeking

NOTE: this is a refinement of the patch from Paul B Mahol
offset all output timestamps by same amount of first input timestamp

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=181031906e4984c1c539dc84d21386a4397e257e
---

 libavfilter/af_atempo.c | 11 ++++++++++-
 1 file changed, 10 insertions(+), 1 deletion(-)

diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c
index bfdad7d76b..688dac5464 100644
--- a/libavfilter/af_atempo.c
+++ b/libavfilter/af_atempo.c
@@ -103,6 +103,9 @@ typedef struct ATempoContext {
     // 1: output sample position
     int64_t position[2];
 
+    // first input timestamp, all other timestamps are offset by this one
+    int64_t start_pts;
+
     // sample format:
     enum AVSampleFormat format;
 
@@ -186,6 +189,7 @@ static void yae_clear(ATempoContext *atempo)
 
     atempo->nfrag = 0;
     atempo->state = YAE_LOAD_FRAGMENT;
+    atempo->start_pts = AV_NOPTS_VALUE;
 
     atempo->position[0] = 0;
     atempo->position[1] = 0;
@@ -1068,7 +1072,7 @@ static int push_samples(ATempoContext *atempo,
     atempo->dst_buffer->nb_samples  = n_out;
 
     // adjust the PTS:
-    atempo->dst_buffer->pts =
+    atempo->dst_buffer->pts = atempo->start_pts +
         av_rescale_q(atempo->nsamples_out,
                      (AVRational){ 1, outlink->sample_rate },
                      outlink->time_base);
@@ -1097,6 +1101,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)
     const uint8_t *src = src_buffer->data[0];
     const uint8_t *src_end = src + n_in * atempo->stride;
 
+    if (atempo->start_pts == AV_NOPTS_VALUE)
+        atempo->start_pts = av_rescale_q(src_buffer->pts,
+                                         inlink->time_base,
+                                         outlink->time_base);
+
     while (src < src_end) {
         if (!atempo->dst_buffer) {
             atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);