mirror of https://github.com/mpv-player/mpv
5fd8a1e04c
This rewrites the audio decode loop to some degree. Audio filters don't do refcounted frames yet, so af.c contains a hacky "emulation". Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of estimating how much audio we need to filter, we always filter full frames. Maybe this should be adjusted later: in case filtering increases the volume of the audio data, we should try not to buffer too much filter output by reducing the input that is fed at once. For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it doesn't seem worth the trouble. |
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.. | ||
af.c | ||
af.h | ||
af_bs2b.c | ||
af_center.c | ||
af_channels.c | ||
af_convert24.c | ||
af_convertsignendian.c | ||
af_delay.c | ||
af_drc.c | ||
af_dummy.c | ||
af_equalizer.c | ||
af_export.c | ||
af_extrastereo.c | ||
af_forcespeed.c | ||
af_format.c | ||
af_hrtf.c | ||
af_hrtf.h | ||
af_karaoke.c | ||
af_ladspa.c | ||
af_lavcac3enc.c | ||
af_lavfi.c | ||
af_lavrresample.c | ||
af_pan.c | ||
af_scaletempo.c | ||
af_sinesuppress.c | ||
af_sub.c | ||
af_surround.c | ||
af_sweep.c | ||
af_volume.c | ||
dsp.h | ||
equalizer.h | ||
filter.c | ||
filter.h | ||
tools.c | ||
window.c | ||
window.h |