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audio: remove sample rate limit checks

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This played the file at a wrong sample rate if the rate was out of
certain bounds.

A comment says this was for the sake of libaf/af_resample.c. This
resampler has been long removed. Our current resampler
(libav/swresample) checks supported sample rates on reconfiguration, and
will error out if a sample rate is not supported. And I think that is
the correct behavior.
This commit is contained in:
wm4 2014-03-30 07:34:43 +02:00
parent 9be2f6b9f8
commit ae448e198f
1 changed files with 1 additions and 7 deletions
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8
player/audio.c
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@ -62,14 +62,8 @@ static int build_afilter_chain(struct MPContext *mpctx)
new_srate = in_format.rate;
else {
new_srate = in_format.rate * opts->playback_speed;
if (new_srate != out_format.rate) {
// limits are taken from libaf/af_resample.c
if (new_srate < 8000)
new_srate = 8000;
if (new_srate > 192000)
new_srate = 192000;
if (new_srate != out_format.rate)
opts->playback_speed = new_srate / (double)in_format.rate;
}
}
return audio_init_filters(d_audio, new_srate,
&out_format.rate, &out_format.channels, &out_format.format);