audio: remove sample rate limit checks

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.

1 files changed, 1 insertions, 7 deletions

diff --git a/player/audio.c b/player/audio.c
index e7c2b8baf4..65c4ad74dd 100644
--- a/player/audio.c
+++ b/player/audio.c
@@ -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);