On 5/8/19 1:13 AM, Paul B Mahol wrote:
On 5/8/19, Pavel Koshevoy <pkoshe...@gmail.com> wrote:
NOTE: this is a refinement of the patch from Paul B Mahol
offset all output timestamps by same amount of first input timestamp
---
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);
--
2.16.4
Should be fine.
Pushed, thank you.
Pavel.
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