audio: cleanup audio filter format negotiation

The algorithm and functionality is the same, but the code becomes much
simpler and easier to follow.

The assumption that there is only 1 conversion filter (lavrresample)
helps with the simplification, but the main change is to use the same
code for format/channels/rate. Get rid of the different AF_CONTROL_SET_*
controls, and change the af->data parameters directly. (af->data is
badly named, but essentially is a placeholder for the output format.)

Also, instead of trying to use the af_reinit() loop to init inserted
conversion filters or filters with changed output formats, do it inline,
and move the common code to a filter_reinit() function. This gets rid of
the awful retry variable.

In general, this should not change any runtime behavior.

1 files changed, 0 insertions, 5 deletions

diff --git a/audio/filter/af.h b/audio/filter/af.h
index 9c49081f66..697024b781 100644
--- a/audio/filter/af.h
+++ b/audio/filter/af.h
@@ -112,9 +112,6 @@ struct af_stream {
 enum af_control {
     AF_CONTROL_REINIT = 1,
     AF_CONTROL_RESET,
-    AF_CONTROL_SET_RESAMPLE_RATE,
-    AF_CONTROL_SET_FORMAT,
-    AF_CONTROL_SET_CHANNELS,
     AF_CONTROL_SET_VOLUME,
     AF_CONTROL_GET_VOLUME,
     AF_CONTROL_SET_PAN_LEVEL,
@@ -160,6 +157,4 @@ int af_test_output(struct af_instance *af, struct mp_audio *out);
 int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
 float af_softclip(float a);
 
-bool af_lavrresample_test_conversion(int src_format, int dst_format);
-
 #endif /* MPLAYER_AF_H */