| Message ID | 20190508045355.12091-1-pkoshevoy@gmail.com |
|---|---|
| State | Accepted |
| Commit | 181031906e4984c1c539dc84d21386a4397e257e |
| Headers | show |
On 5/8/19, Pavel Koshevoy <pkoshevoy@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 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". Should be fine.
On 5/8/19 1:13 AM, Paul B Mahol wrote: > On 5/8/19, Pavel Koshevoy <pkoshevoy@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.
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);