| Message ID | 20190124124203.19345-1-pawel.wegner95@gmail.com |
|---|---|
| State | New |
| Headers | show |
On 1/24/19, Paweł Wegner <pawel.wegner95@gmail.com> wrote: > Signed-off-by: Paweł Wegner <pawel.wegner95@gmail.com> > --- > libavfilter/af_atempo.c | 9 +++++++++ > 1 file changed, 9 insertions(+) > > diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c > index bfdad7d76b..1245eae8c1 100644 > --- a/libavfilter/af_atempo.c > +++ b/libavfilter/af_atempo.c > @@ -147,6 +147,8 @@ typedef struct ATempoContext { > uint8_t *dst_end; > uint64_t nsamples_in; > uint64_t nsamples_out; > + > + int64_t first_frame_pts; > } ATempoContext; > > #define YAE_ATEMPO_MIN 0.5 > @@ -994,6 +996,7 @@ static av_cold int init(AVFilterContext *ctx) > ATempoContext *atempo = ctx->priv; > atempo->format = AV_SAMPLE_FMT_NONE; > atempo->state = YAE_LOAD_FRAGMENT; > + atempo->first_frame_pts = AV_NOPTS_VALUE; > return 0; > } > > @@ -1069,6 +1072,7 @@ static int push_samples(ATempoContext *atempo, > > // adjust the PTS: > atempo->dst_buffer->pts = > + (atempo->first_frame_pts == AV_NOPTS_VALUE ? 0 : > atempo->first_frame_pts / atempo->tempo) + > av_rescale_q(atempo->nsamples_out, > (AVRational){ 1, outlink->sample_rate }, > outlink->time_base); > @@ -1108,6 +1112,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame > *src_buffer) > > atempo->dst = atempo->dst_buffer->data[0]; > atempo->dst_end = atempo->dst + n_out * atempo->stride; > + > + if (atempo->first_frame_pts == AV_NOPTS_VALUE) > + atempo->first_frame_pts = > av_rescale_q(atempo->dst_buffer->pts, > + inlink->time_base, > + outlink->time_base); > } > > yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end); > -- > 2.17.1 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > lgtm
On Thu, Jan 24, 2019 at 5:49 AM Paweł Wegner <pawel.wegner95@gmail.com> wrote: > > Signed-off-by: Paweł Wegner <pawel.wegner95@gmail.com> > --- > libavfilter/af_atempo.c | 9 +++++++++ > 1 file changed, 9 insertions(+) > > diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c > index bfdad7d76b..1245eae8c1 100644 > --- a/libavfilter/af_atempo.c > +++ b/libavfilter/af_atempo.c > @@ -147,6 +147,8 @@ typedef struct ATempoContext { > uint8_t *dst_end; > uint64_t nsamples_in; > uint64_t nsamples_out; > + > + int64_t first_frame_pts; > } ATempoContext; > > #define YAE_ATEMPO_MIN 0.5 > @@ -994,6 +996,7 @@ static av_cold int init(AVFilterContext *ctx) > ATempoContext *atempo = ctx->priv; > atempo->format = AV_SAMPLE_FMT_NONE; > atempo->state = YAE_LOAD_FRAGMENT; > + atempo->first_frame_pts = AV_NOPTS_VALUE; > return 0; > } > > @@ -1069,6 +1072,7 @@ static int push_samples(ATempoContext *atempo, > > // adjust the PTS: > atempo->dst_buffer->pts = > + (atempo->first_frame_pts == AV_NOPTS_VALUE ? 0 : atempo->first_frame_pts / atempo->tempo) + > av_rescale_q(atempo->nsamples_out, > (AVRational){ 1, outlink->sample_rate }, > outlink->time_base); > @@ -1108,6 +1112,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer) > > atempo->dst = atempo->dst_buffer->data[0]; > atempo->dst_end = atempo->dst + n_out * atempo->stride; > + > + if (atempo->first_frame_pts == AV_NOPTS_VALUE) > + atempo->first_frame_pts = av_rescale_q(atempo->dst_buffer->pts, > + inlink->time_base, > + outlink->time_base); > } > > yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end); > -- > 2.17.1 > Probably okay. The reason I didn't do this to begin with is because this is an audio stream filter... and how the timeline of the stream was transformed up to the 1st frame is unknown. You are making the assumption that it should have been transformed using the same tempo parameter as current tempo, but (video) tempo can be varied at runtime prior to 1st audio frame, so pts_0' = pts_0 / tempo could be wrong. Anyway, I don't have a use case where this change would break something, so if this fixes something for you then it's fine. Pavel.
On Thu, Jan 24, 2019 at 3:43 PM Pavel Koshevoy <pkoshevoy@gmail.com> wrote: > On Thu, Jan 24, 2019 at 5:49 AM Paweł Wegner <pawel.wegner95@gmail.com> > wrote: > > > > Signed-off-by: Paweł Wegner <pawel.wegner95@gmail.com> > > --- > > libavfilter/af_atempo.c | 9 +++++++++ > > 1 file changed, 9 insertions(+) > > > > diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c > > index bfdad7d76b..1245eae8c1 100644 > > --- a/libavfilter/af_atempo.c > > +++ b/libavfilter/af_atempo.c > > @@ -147,6 +147,8 @@ typedef struct ATempoContext { > > uint8_t *dst_end; > > uint64_t nsamples_in; > > uint64_t nsamples_out; > > + > > + int64_t first_frame_pts; > > } ATempoContext; > > > > #define YAE_ATEMPO_MIN 0.5 > > @@ -994,6 +996,7 @@ static av_cold int init(AVFilterContext *ctx) > > ATempoContext *atempo = ctx->priv; > > atempo->format = AV_SAMPLE_FMT_NONE; > > atempo->state = YAE_LOAD_FRAGMENT; > > + atempo->first_frame_pts = AV_NOPTS_VALUE; > > return 0; > > } > > > > @@ -1069,6 +1072,7 @@ static int push_samples(ATempoContext *atempo, > > > > // adjust the PTS: > > atempo->dst_buffer->pts = > > + (atempo->first_frame_pts == AV_NOPTS_VALUE ? 0 : > atempo->first_frame_pts / atempo->tempo) + > > av_rescale_q(atempo->nsamples_out, > > (AVRational){ 1, outlink->sample_rate }, > > outlink->time_base); > > @@ -1108,6 +1112,11 @@ static int filter_frame(AVFilterLink *inlink, > AVFrame *src_buffer) > > > > atempo->dst = atempo->dst_buffer->data[0]; > > atempo->dst_end = atempo->dst + n_out * atempo->stride; > > + > > + if (atempo->first_frame_pts == AV_NOPTS_VALUE) > > + atempo->first_frame_pts = > av_rescale_q(atempo->dst_buffer->pts, > > + > inlink->time_base, > > + > outlink->time_base); > > } > > > > yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end); > > -- > > 2.17.1 > > > > > Probably okay. The reason I didn't do this to begin with is because > this is an audio stream filter... and how the timeline of the stream > was transformed up to the 1st frame is unknown. You are making the > assumption that it should have been transformed using the same tempo > parameter as current tempo, but (video) tempo can be varied at runtime > prior to 1st audio frame, so pts_0' = pts_0 / tempo could be wrong. > > Anyway, I don't have a use case where this change would break > something, so if this fixes something for you then it's fine. This fixes seeking when I have video playback sped up in ffplay like this: ffplay -vf "setpts=0.5 * PTS" -af "atempo=2" input > > Pavel. > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel >
On 24-01-2019 08:18 PM, Paweł Wegner wrote: > > This fixes seeking when I have video playback sped up in ffplay like this: > ffplay -vf "setpts=0.5 * PTS" -af "atempo=2" input The better way to run this is ffplay -vf "setpts=0.5 * (PTS - STARTPTS)" -af "atempo=2" input Gyan
On Thu, Jan 24, 2019 at 8:01 AM Gyan <ffmpeg@gyani.pro> wrote: > > > > On 24-01-2019 08:18 PM, Paweł Wegner wrote: > > > > This fixes seeking when I have video playback sped up in ffplay like this: > > ffplay -vf "setpts=0.5 * PTS" -af "atempo=2" input > > The better way to run this is > > ffplay -vf "setpts=0.5 * (PTS - STARTPTS)" -af "atempo=2" input > > yeah, why not just do this... no code changes required. I'd prefer it if atempo stayed simple. This way you can still use a separate pts transform filter to adjust audio pts if necessary and without hard-coding an assumption into atempo. Pavel.
On Thu, Jan 24, 2019 at 4:01 PM Gyan <ffmpeg@gyani.pro> wrote: > > > On 24-01-2019 08:18 PM, Paweł Wegner wrote: > > > > This fixes seeking when I have video playback sped up in ffplay like > this: > > ffplay -vf "setpts=0.5 * PTS" -af "atempo=2" input > > The better way to run this is > > ffplay -vf "setpts=0.5 * (PTS - STARTPTS)" -af "atempo=2" input > This works fine, thanks; the patch can be discarded. > > Gyan > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel >
diff --git a/libavfilter/af_atempo.c b/libavfilter/af_atempo.c index bfdad7d76b..1245eae8c1 100644 --- a/libavfilter/af_atempo.c +++ b/libavfilter/af_atempo.c @@ -147,6 +147,8 @@ typedef struct ATempoContext { uint8_t *dst_end; uint64_t nsamples_in; uint64_t nsamples_out; + + int64_t first_frame_pts; } ATempoContext; #define YAE_ATEMPO_MIN 0.5 @@ -994,6 +996,7 @@ static av_cold int init(AVFilterContext *ctx) ATempoContext *atempo = ctx->priv; atempo->format = AV_SAMPLE_FMT_NONE; atempo->state = YAE_LOAD_FRAGMENT; + atempo->first_frame_pts = AV_NOPTS_VALUE; return 0; } @@ -1069,6 +1072,7 @@ static int push_samples(ATempoContext *atempo, // adjust the PTS: atempo->dst_buffer->pts = + (atempo->first_frame_pts == AV_NOPTS_VALUE ? 0 : atempo->first_frame_pts / atempo->tempo) + av_rescale_q(atempo->nsamples_out, (AVRational){ 1, outlink->sample_rate }, outlink->time_base); @@ -1108,6 +1112,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer) atempo->dst = atempo->dst_buffer->data[0]; atempo->dst_end = atempo->dst + n_out * atempo->stride; + + if (atempo->first_frame_pts == AV_NOPTS_VALUE) + atempo->first_frame_pts = av_rescale_q(atempo->dst_buffer->pts, + inlink->time_base, + outlink->time_base); } yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);
Signed-off-by: Paweł Wegner <pawel.wegner95@gmail.com> --- libavfilter/af_atempo.c | 9 +++++++++ 1 file changed, 9 insertions(+)