obs-ffmpeg: Use actual audio encoder sample rate

Uses the sample rate the audio encoder is running at for ffmpeg muxing
(in case the audio sample rate had to be changed by the encoder)

plugins/obs-outputs/flv-mux.c

@@ -95,7 +95,7 @@ static bool build_flv_meta_data(obs_output_t *context,
 	enc_str_val(&enc, end, "audiocodecid", "mp4a");
 	enc_num_val(&enc, end, "audiodatarate", encoder_bitrate(aencoder));
 	enc_num_val(&enc, end, "audiosamplerate",
-			(double)audio_output_get_sample_rate(audio));
+			(double)obs_encoder_get_sample_rate(aencoder));
 	enc_num_val(&enc, end, "audiosamplesize", 16.0);
 	enc_num_val(&enc, end, "audiochannels",
 			(double)audio_output_get_channels(audio));