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| author | wm4 <wm4@nowhere> | 2020-05-23 04:04:46 +0200 |
|---|---|---|
| committer | wm4 <wm4@nowhere> | 2020-05-23 04:04:46 +0200 |
| commit | ab4e0c42fbb11f2e7459fe9fe98265e0323b1ef4 (patch) | |
| tree | 8d202d9a8d7aecf3f305c0311d5115b46ba009d9 /filters/user_filters.c | |
| parent | 43a67970b6c17f8d0525f90fac25923cd4a989bf (diff) | |
| download | mpv-ab4e0c42fbb11f2e7459fe9fe98265e0323b1ef4.tar.bz2 mpv-ab4e0c42fbb11f2e7459fe9fe98265e0323b1ef4.tar.xz | |
audio: redo video-sync=display-adrop
This mode drops or repeats audio data to adapt to video speed, instead
of resampling it or such. It was added to deal with SPDIF. The
implementation was part of fill_audio_out_buffers() - the entire
function is something whose complexity exploded in my face, and which I
want to clean up, and this is hopefully a first step.
Put it in a filter, and mess with the shitty glue code. It's all sort of
roundabout and illogical, but that can be rectified later. The important
part is that it works much like the resample or scaletempo filters.
For PCM audio, this does not work on samples anymore. This makes it much
worse. But for PCM you can use saner mechanisms that sound better. Also,
something about PTS tracking is wrong. But not wasting more time on
this.
Diffstat (limited to 'filters/user_filters.c')
| -rw-r--r-- | filters/user_filters.c | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/filters/user_filters.c b/filters/user_filters.c index 72a2ab892c..57021adc7c 100644 --- a/filters/user_filters.c +++ b/filters/user_filters.c @@ -39,6 +39,7 @@ const struct mp_user_filter_entry *af_list[] = { &af_rubberband, #endif &af_lavcac3enc, + &af_drop, }; static bool get_af_desc(struct m_obj_desc *dst, int index) |